hi friends,
I tried to convert mkv movie into xvid using megui and i have some doubts in lame mp3 audio settings. I set lame mp3 "scratchpad" ->sample rate keep original -> ABR 80 in MeGUI (pls check IMG 3)
I can't understand that why my Converted movie info settings always shows q2 like this [Encoding settings: -m j -V 4 -q 2 -lowpass 13.5 -- abr 80] and sample rate automatically changed into 32.0 KHZ even i set it to keep original sample rate (pls check IMG 2)
Most of the movie uploaders file info shows q3 like this [Encoding settings : -m j -V 4 -q 3 -lowpass 13.5 -- abr 80] and sample rate shows 48.0 KHZ (pls check IMG 1)
and i hope you will understand what i am trying to say. Anyway to set the audio settings like the movie uploaders doing there. pls give me some advice thanks...
IMG 1
IMG 2
IMG 3
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Look at your IMG 3 above. see the "extra" tab? You'll find a custom command line, add it there
Why is -q 3 so important? -
Yeah custom command line is there but how to type commands pls give any samples. q3 is not so important I just want to convert movies like torrent uploaders. I want to know how they set sampling rate 48.0 khz and
Encoding settings : -m j -V 4 -q 3 -lowpass 13.5 -- abr 80 like this.
I tried to convert with the same marked settings like the (IMG 1) shows here but the converted output file info shows 32.0 khz sampling rate even i set it to 48.0 khz sampling rate. Pls compare IMG 1 & IMG 2 for better understanding. Here Img1 shows torrent uploaders settings and img 2 shows my converted file info. -
Save this file as LAME MP3_dp_ sam78.xml at Megui > allprofiles > LAME MP3
Code:<?xml version="1.0"?> <GenericProfileOfMP3Settings xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance" xmlns:xsd="http://www.w3.org/2001/XMLSchema"> <Name>sam78</Name> <Settings> <PreferredDecoderString>LWLibavAudioSource</PreferredDecoderString> <DownmixMode>StereoDownmix</DownmixMode> <BitrateMode>ABR</BitrateMode> <Bitrate>128</Bitrate> <AutoGain>true</AutoGain> <SampleRateType>deprecated</SampleRateType> <SampleRate>KeepOriginal</SampleRate> <TimeModification>KeepOriginal</TimeModification> <ApplyDRC>false</ApplyDRC> <Normalize>100</Normalize> <CustomEncoderOptions>-m j -V 4 -q 3 --lowpass 13.5 --abr 80 --resample 48</CustomEncoderOptions> <Quality>4</Quality> <AbrBitrate>128</AbrBitrate> </Settings> </GenericProfileOfMP3Settings>
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@amaipaipai Nice explaination thank you so much. I have some doubts! If we want to change audio bitrate 96 or 128 or sample rate 41.0 khz or 48.0 khz then we need to edit that xml commands every time or we can make changes directly using megui config button that we can see in the above image right side, marked with red line.
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The -q setting sets the quality of the encoding algorithm. The lower the number, the higher the quality, even for constant bitrate encoding, but it also takes longer.
-q 0 is the best. It's also so slow it's almost unusable.
-q 2 was once the default for constant/average bitrate encoding, but it was changed to -q 3 a long time ago, however as -q 2 is only a bit slower and in theory better quality, some encoder GUI's stuck with -q 2. MeGUI is one of them.
There's nothing wrong with MeGUI's resampling. The problem is the LAME encoder is trying to punish you for using too low a bitrate, so to keep the quality as high as possible it's automatically reducing the sample rate to 32k. If you changed the bitrate to 128 (or maybe something in between 80 and 128) the encoded MP3 will in fact be 48k (if you use MeGUI's resampling function to change it to 48000 Hz if it's not 48k already).
From the log file, under "Standard Error Stream":
[Information] [21/02/18 9:38:15 PM] Standard error stream
-[Information] [21/02/18 9:38:25 PM] LAME 3.100 32bits (http://lame.sf.net)
-[Information] [21/02/18 9:38:25 PM] CPU features: MMX (ASM used), SSE (ASM used), SSE2
-[Information] [21/02/18 9:38:25 PM] Resampling: input 48 kHz output 32 kHz
-[Information] [21/02/18 9:38:25 PM] Using polyphase lowpass filter, transition band: 13548 Hz - 13935 Hz
-[Information] [21/02/18 9:38:25 PM] Encoding <stdin> to D:\Need More Bitrate.mp3
-[Information] [21/02/18 9:38:25 PM] Encoding as 32 kHz j-stereo MPEG-1 Layer III (12.8x) average 80 kbps qval=2
If you know better than the encoder, you can force 48k while using a low bitrate by adding:
--resample 48
to the custom command line section under the Extras tab. That'll force the encoder to keep the sample rate at 48k, or convert it to 48 if it's not. Don't do it though. Increase the bitrate instead.
Edit: I just noticed that's what the xml file posted by amaipaipai does, it adds --resample 48 to the command line. I'd still vote for not doing it though and increasing the bitrate instead.Last edited by hello_hello; 21st Feb 2018 at 06:22.
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The xml is the settings saved from what you define on MeGUI screen, all you have to do is to change whatever you need and save it to keep it. The xml is just a guide so you can see on MeGUI what I've done, you can change to whatever settings you see fit to your needs.
I don't agree with any of that settings myself, I'm not a MeGUI user, all I need to do I script with avisynth and other manual tools.
Xvid is a obsolete codec, a lowpass of 13.5 makes no sense at all to me, maybe this is the player requirements he uses, don't know. With so many "wrongs" the resample settings will make no difference, since he needs a low bitrate (?) and high samplerate (?) the settings will meet his needs. -
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The way I'd phrase the question would be, "will 128k decrease the quality less than 80k?" If that's the question, then the answer is yes. The bitrate of the source audio is somewhat irrelevant. The original audio is decoded as a wave file and sent to the MP3 encoder for encoding. The encoder is oblivious to anything else. It just sees a wave file whether the source was 1536k DTS or 64k AAC, and the quality at which it'll encode that wave file is primarily determined by the bitrate.
Why 80k anyway? If you work it out, 2 hours of audio at 80k results in an MP3 that's roughly 70MB in size (if I got the maths right). At 128k it'd be 112MB. Even at 192k it's still only 169MB, but over two hours you only gain 42MB by increasing the bitrate to 128k, which is almost nothing considering video files are often gigabytes in size.
If you're encoding for a particular file size, then you can decrease the video bitrate to compensate. Reducing the bitrate by 48k for the video wouldn't have a noticeable effect on quality.... unless you're already using very low bitrates. -
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My bad on this comment. I did not fully understand that you were talking about a source of 80kbit. If you re-encode this 80kbit audio file, you won't get any quality back as any losses are already gone. Whether the losses be MP3 artifacts, a lower sample rate (32khz), or having the higher frequencies cut off because LAME MP3 is trying to save bandwidth.
My "yes" comment was to say that encoding with 128kbit would probably give better audio results than 80kbit, as it will keep more of the source audio quality. Encoding with 160kbit-192kbit LAME MP3 would be more ideal and near transparent. -
Friends here I saw your comments really each and every comments are so helpful for me thanks a lot friends.
This is my last question friends try to answer me if it is possible.. thanks in advance.
Have you ever watched a movie where the dialogue was almost inaudible, but then all of a sudden an action scene comes in and shakes your house with noise?
Anyway to fix that using Megui. I want to encode the whole movie in same volume level (dialogues and action scene should be same volume level) Pls help me friends... -
You can't do it with MeGUI as such (edit, there's a way, see below), but you can do it with foobar2000. It has DSPs for decoding the formats (DTS and AC3 etc) it can't decode "out of the box".
For compressing when converting I use ffmpeg as the encoder with the built in Dynamic Audio Normaliser compressing the audio. foobar2000 has a DSP for down-mixing to stereo if required. It'll involve a bit of a learning curve to set it up if you've never used foobar2000 before.
If you want to try it, this is an encoder configuration for converting to 128k ABR MP3 while compressing with the Dynamic Audio Normaliser.
Edit. Thinking about it, you can do it with MeGUI as long as you convert to MP2 or AC3 because MeGUI uses ffmpeg to encode the audio for those formats. You might want to test it out first before diving into using foobar2000. Select AC3 or MP2 as the format and add the following to the custom command line section of the encoder configuration.
-af dynaudnorm=f=150
MeGUI badly needs an encoder configuration you can set up from scratch using any custom command line encoder.
The foobar2000/ffmpeg command line:
-i - -ignore_length true -af dynaudnorm=f=150 -c:a libmp3lame -b:a 128k -abr 1 %dLast edited by hello_hello; 3rd Mar 2018 at 02:45.
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@hello_hello bro I think you are master in encoding thats nice. Yes u r right bro I never used foobar2000 but I want to learn command line encoding like you have done in above image so please give me some tutorial and how can I get the screen like the above IMG. Pls guide me step by step.. thanks
Last edited by sam78; 3rd Mar 2018 at 09:28.
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I'd have to write an essay to give step by step instructions. Install it, play around, and if there's anything you need help with, ask specific questions. Someone will be able to help.
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@hello_hello
Can i use the below encoder configuration for converting to 96k CBR MP3 while compressing with the Dynamic Audio Normaliser.
The foobar2000/ffmpeg command line:
-i - -ignore_length true -af dynaudnorm=f=150 -c:a libmp3lame -b:a 96k -cbr 1 %d
I guess 150 is default value am i right? So we never change this?
-af dynaudnorm=f=150 -
@sneaker thanks for the info friend. You have any idea why we use this value 150 -af dynaudnorm=f=150
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