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23 replies to this topic

#1
Cman21

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here is a simple tool made by Qriist on the OZCrew (OZC Staff) that will encode virtually any audio format into AAC and if stereo AAC HEv2 for even better compression with virtually no fidelity loss. by default the transmux.ini file has the compression set to 1.0 for best quality but if you want better compression with less quality you can lower this number to about .5 before it starts to sound really bad.

using this tool is very simple.
  • put all the audio files you want converted into the "Input Non-AAC Source" folder.
  • run "AAC.exe" in the main folder
  • delete old source files from "Output Source" folder, or do whatever you want with the source files
  • enjoy your new AAC files in the "Output Transmux" folder.
so now that you have the run down here is a link to the file:
https://mega.co.nz/#!FRY3AJjB!TS0nsCLn_NPwp8t-0b7bbA3p-QOqJ-Tp-hBXef_5D9Y
if the link ever goes offline just shoot me a PM

#2
JohnFlower

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Oh, thanks for this. Although I have a relatively high-end headphone, I really can't tell the difference between FLAC and AAC.

#3
Megagold5

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Looks interesting, shall have to try it out at some point. Thanks.

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#4
hhsquall

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Oh, thanks for this. Although I have a relatively high-end headphone, I really can't tell the difference between FLAC and AAC.


True! As a matter of fact, no one can. In terms of reconstruction of the original sound, only a digital comparision can verify the difference between a lossy track, and a lossless one. Human's ears are incapable of the discrimination, even with the highest-end equipment (headphones, speakers, etc.).
Unfortunately, this has become a trend for most releasers to go with the FLAC, which eats up resources (bandwidth and storage for both the distributors and users).
Personally, I don't have the time, and of course raw materials, to do any encoding, but as the Cman kindly provided here, try and re-encode everything I have in FLAC, into AAC (and keep only the Japanese audio except for very few animes).
I am an audiophile and still find it unreasonable... However, as I am actually downloading other people's work, I don't complain on their method, and simply look for the best possible option :)
Cheers :lol:

#5
Pooba

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uummm......i really dont know how you cant tell the different between flac and aac.when you encode to aac mids and lows are reduced alot.i cant see how you say you have the highest end epuipment and cant tell the different,im using 7.1ch dss system and headsets,my head sets cost me about 300 and i even use a mixer with it and i can tell the difference between the 2 with standard head sets.

#6
Cman21

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being able to hear the difference in AAC and FLAC really comes down to 3 factors.

  • your own personal hearing and age. it is proven that the younger you are the better your hearing is, if you do not do something to hurt your hearing.
  • your equipment, you can not use cheap $10 speakers from walmart and think FLAC will help you. you must have a high fidelity sound card with a great headset. you could have high end speakers but due to ambient noises it makes it harder to hear the differences.
  • and the one factor that matters most is the quality factor the encoder selects to compress to. everyone will hear the difference even on those walmart speakers if the encoder drops the quality factor on the AAC compression down to 50%. but if kept at 100% the factors above are what truly matter.


#7
hhsquall

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@Cman:
Duly noded.
I forgot to mention the compression method and ratio; of course almost anyone can discern a bad lossy compression from a lossless one :)

@Pooba:
True, but as Cman mentioned, there are multiple factors contributing to that. It is not a clear-cut threshold; compression settiings can make a huge difference.

Also, I separate anime from general music listening. I am really picky on the latter and would go with the lossless of course!

#8
Koby

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What would we edit to change the settings?


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#9
Forge

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There's an .ini file that changes the Q[uality] of the audio. I don't think there's away to change it to CBR or ABR.

Also, check to make sure the AAC .exe you have is the newest one.

#10
Cman21

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yup the transmux.ini file in your main transmux folder which you can open in notepad or something and then edit the quality setting there. that is the only thing you can change settings wise, everything else is automatic.

#11
Hark0n

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Looked interesting, so I did a small test:
1. Is seems to be unable to open AIFF, ALAC, APE, MPC, NeroAAC, TTA and WMA formats. At least on settings I encoded them.
2. It shifted audio 100ms forward for some reason (converting WAV to AAC). It is possible that other encoder I used shifted audio and so it went wrong, but it is less likely.
3. NeroAAC at about the same filesize produced track more closer to source (judging from graphical representation). The sound might be better but it is hard to tell simply by listening. Due to shifting null test failed and I was too lazy to try and fix it.

Considering it is a free tool, it did decent job. Also it is easy to use end configure.

Also setting quality value to a number above 1, it created 3 files with different bitrates - "name (HEv1).mp4", "name (HEv2).mp4" and "name (LC).mp4".

#12
Cman21

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didnt know that it would go through all the compression methods by setting it higher than 1... but i do love Hev2 for stereo tracks as it compresses very well. i did know that it wouldnt support ALAC, as that is an apple thing, and it doesnt re-encode AAC. but i didnt know that it wouldnt support wma... oh well, for those other audio types you can always throw it into an audio editor like audacity and export as a wav.

#13
Hark0n

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Audio shifting is presented when converting all formats (AC3, FLAC, MP2, MP3, OGG, WAV, WavPack). In LC it is ~30ms. Any idea what is causing it?
Also HEv1 and LC is noticeably closer to source comparing to HEv2. As I mentioned previously, quality value can be set higher than 1, but there is no difference from 2 or 20, so no point of changing it too high.
I didn't test this on DTS, DTS-MA and TrueHD, since I don't have converters for them.

#14
Cman21

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i never noticed any shifting messing up sync back when i used to use this tool. but i have no idea why or how it could be shifting the audio.

#15
Hark0n

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i never noticed any shifting messing up sync back when i used to use this tool. but i have no idea why or how it could be shifting the audio.

The shifting is less than 100ms (one tenth of a second), so it is hard to spot. I spotted it in audio editor not in playback. It seems to be shifted by encoder itself (neroAacEnc.exe) not Transmux. I used this encoder with different GUI, and got ~35ms shifting.
Changing encoder removes shifting in LC, but it still remains in HEv1 and HEv2.

#16
Cman21

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hmm ya i tend to be good at noticing audio miss sync as little as 50ms but i guess it wasnt enough for me to look for it. that or the shift is coming in play due to the container and is removed when muxed into an mkv container, can you test thar theory please?

#17
Hark0n

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hmm ya i tend to be good at noticing audio miss sync as little as 50ms but i guess it wasnt enough for me to look for it. that or the shift is coming in play due to the container and is removed when muxed into an mkv container, can you test thar theory please?

As far as mkv goes, it doesn't seem to be the case, but the mp4 file contains chapter which cuts of 127ms (in my case) so that audio shifting is nullified.
Since most audio editors can't open AAC, I am converting files to WAV. It is possible that converting shifts audio forward by chapters length. By cutting beginning of the file (part before chapter) shifting can be fixed, more or less.
This is not the only audio converter that shifts audio. Even some commercial converters does the same.

#18
Cman21

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interesting, thanks for your in depth analysis.

#19
DeathScythe

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Thanks for the tool, very simple and easy to use. Just finished converting a flac track to aac and it did a nice job.

#20
Evil_Burrito

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Oh schnap cman got a shout out on that page.

Thanks, I will give the utility a test. Currently I use LameXP for 95% of my audio encoding.

btw, is there a reason the program uses neroAacEnc1.3.3.0 instead of neroAacEnc 1.5.4.0? Can/should I just replace it with the newer version?


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