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1
3rd Party Plugins - (fb2k) / Re: SACD .dsf file conversion plug-ins
Last post by Porcus -
the "multichannel" WAV files are significantly larger than the equivalent stereo WAV files
Yes, more channels take more bits to encode.

which are significantly larger than the equivalent WAV files of the equivalent MFSL standard CD.
Because DSD streams are not PCM (they are encoded according to different principles - DSD are more like how a "dimmer" on a light quickly switches on and off yet produces an intermediate brightness level). So in order to capture as much as possible, that sample rate, the files become enormous.

The CD could also be of a different mastering.[/quote]



One thing I did note was that FB2k kept telling me that the conversion process was not "lossless", and I did notice a somewhat smaller file size for a converted WAV file than its source DSF file. 
That is strange - the WAV would normally be much larger. (Are you comparing the multichannel?)

But the conversion is not lossless because the files are originally encoded according to totally different principles. DSD streams in DSF files are one bit but extremely high sample rate (again, like a dimmer) - PCM streams in WAV (or AIFF or FLAC) are a number of bits and sampled at an octave over the highest note.

I'm guessing that I am not personally going to notice any difference in audio quality.
You won't. fb2k converts at 32-bit accuracy, and you wouldn't have noticed any difference at half of that. And that is only in the bit depth - also it converts octaves above the audible range.
For two channels you wouldn't notice at the file size of the CD either - SACD was a nonsense format for stereo.

You can find some recordings in various formats at http://www.2l.no/hires/ . To give you an idea of sizes, consider "Finzi: Come Away, Death", stereo only
* 11 megabytes for CD format, FLAC.
* 37 for CD format, WAV
* 55 for 96/24, FLAC
* 69 for DSD packed in WavPack
* 110 for DSD as .dsf - that is ten times as much as the CD format FLAC.
* 442 for DSD converted to WAV.

You won't hear any difference between the 11 megabytes file and one that is forty times the size.
2
Listening Tests / Re: Personal blind sound quality comparison of Opus hard-CBR with framesize options
Last post by Kamedo2 -
On the applaud sample, the first few seconds are interpreted as speech, and Silk apparently sounds quite a bit worse on applause-like signals than CELT.

C.R.Helmrich, thank you for the clear analysis.
Indeed, the first two seconds or so sounds like Silk-ish (the applaud is muffled and muted), and the rest sounds like CELT-ish (vivid and bright), and the abrupt increase in volume alters the nuances of the scene.
Forcing the use of CELT with --music indeed solves the problem.
Now, all of those applauds below sounds lively and close to the original.

opus-tools-0.2-opus-1.3\opusenc --hard-cbr --bitrate 48 --music --framesize 20 "07 applaud.wav" "07 applaud.cbr.music.48k.opus"
opus-tools-0.2-opus-1.3\opusenc --hard-cbr --bitrate 52 --music --framesize 20 "07 applaud.wav" "07 applaud.cbr.music.52k.opus"
opus-tools-0.2-opus-1.3\opusenc --hard-cbr --bitrate 56 --music --framesize 20 "07 applaud.wav" "07 applaud.cbr.music.56k.opus"
opus-tools-0.2-opus-1.3\opusenc --hard-cbr --bitrate 60 --music --framesize 20 "07 applaud.wav" "07 applaud.cbr.music.60k.opus"
opus-tools-0.2-opus-1.3\opusenc --bitrate 48 --music --framesize 20 "07 applaud.wav" "07 applaud.music.48k.opus"
opus-tools-0.2-opus-1.3\opusenc --bitrate 52 --music --framesize 20 "07 applaud.wav" "07 applaud.music.52k.opus"
opus-tools-0.2-opus-1.3\opusenc --bitrate 56 --music --framesize 20 "07 applaud.wav" "07 applaud.music.56k.opus"
opus-tools-0.2-opus-1.3\opusenc --bitrate 60 --music --framesize 20 "07 applaud.wav" "07 applaud.music.60k.opus"
3
General Audio / Re: Detecting whether a 24-bit file has been upconverted from 16-bit?
Last post by bennetng -
The distribution of samples are fine, no "hint" of simple upconversion within what the software is able to detect.

The index of bits is an abstraction of audio level, the top (0) denotes silence, the bottom (23) denotes samples located at the highest 6.0206dB, and every lower index denotes the subsequent 6.0206dB.

If you look at the examples I included in the readme file, you can see that in the electronic music album, there are more samples located at the higher bits and they are packed more tightly within a fewer number of bits. On the other hands, the classical album has fewer samples located at the highest bits, and the values spanned more sparsely among more bits. Which means the classical album has a higher requirement of bit-depth than the electronic album.

Your file is somewhere between my two examples, which means the bit-depth requirement is not particularly high.
4
WavPack / Re: How to properly pack DSF files into WavPack?
Last post by bryant -
Not exactly an "odd" format, but that file has ID3v2.4 tags, not ID3v2.3, which is the only version WavPack currently supports. Adding v2.4 is definitely on my list of todo's, but the specs are so bad and I've heard so many horror stories that I'm a little hesitant.

But it is on the list...  :)
5
General Audio / Re: Equalise Volumes via dbPoweramp (-db) & wxMp3gain (+db) conversion
Last post by DVDdoug -
Quote
But in your example, isn't foobar going to increase all tracks by +4db, rather than a target db level?
He said, "+4 LUFS to target".

Quote
Also, wxmp3gain is documented to make lossless edits, by altering the volume on the actual file without transcoding. Are you sure it's not lossless?
I agree,   It's only a loudness change and it's reversible.

Technically, when you change the volume of a WAV file there are rounding errors and when you reduce the volume you loose resolution.   But it's not considered to be a lossy process.   Mixing & mastering engineers make all kinds of volume adjustments without worrying about if it's "mathematically perfect" or "mathematically reversible"

Quote
Also, how are you calculating ReplayGain to New ReplayGain? Is there a chart, or conversion table, for each number?
I've done an experiment and I don't remember the results but I'll trust Markuza that there's a 107dB difference between the Acoustic SPL level and the digital LUFS level.    I assume this is with pink noise.   With real music there isn't an exact correlation because they use different equal loudness curves and/or a different reference SPL level and it will depend on the frequency content.

Quote
I want to achieve either 75db, -14 LUFS (Spotify standard), or 89db.
I haven't decided which one because I want to allow enough headroom for clipping and listen to all genres of music.
It's a compromise and a lot of people complain that 89dB is "too quiet".    On the other hand if you go to 93dB a lot of your tracks won't be touched unless you allow clipping.    At -75dB you might still have a few tracks that can't hit the target loudness if you don't allow clipping.     You're also never going to get "perfect" loudness matching because your brain doesn't do this kind of "analysis".    You might have one song that starts-soft and ends loud or vice-versa and your brain/perception will do "funny things".   Or you might "hear" heavy metal louder than classical when they are really the same volume.   Two different people might not agree when two different songs sound equally loud.

MP3 can go over 0dB without clipping but you can still clip your DAC if you play it at "full digital volume".    I'm not sure what the upper limit is and it might not be a "simple-fixed" limit.
6
General Audio / Re: Equalise Volumes via dbPoweramp (-db) & wxMp3gain (+db) conversion
Last post by hmp -
I cannot edit my previous post. Here's a visual guide how to achieve what you are looking for.

Spoiler (click to show/hide)

Thank you for explaining!
But in your example, isn't foobar going to increase all tracks by +4db, rather than a target db level?
Also, wxmp3gain is documented to make lossless edits, by altering the volume on the actual file without transcoding. Are you sure it's not lossless?
Also, how are you calculating ReplayGain to New ReplayGain? Is there a chart, or conversion table, for each number?
7
General Audio / Re: Equalise Volumes via dbPoweramp (-db) & wxMp3gain (+db) conversion
Last post by Markuza97 -
I cannot edit my previous post. Here's a visual guide how to achieve what you are looking for.

Spoiler (click to show/hide)
8
General Audio / Re: Detecting whether a 24-bit file has been upconverted from 16-bit?
Last post by Abstracter -
I've played-around with some of these tools in the past and they were easy to fool and I recently downloaded Bitter (which I tried in Audacity).     MP3s from ripped CDs show about 30-bits (which is true depending on the MP3 decoder).
bitter can't even survive simple dithering, let alone mp3.
https://hydrogenaud.io/index.php?topic=114816.msg992983#msg992983
My software cannot survive lossy compression either, but for basic operations like volume adjustment and dithering, it works.

I used your BitSort on some files I suspect were upconverted and got this

00:03:24.2266666 = 18012792 samples / 2-ch @ 44100Hz
24-bit fixed point
Bit   Count         Percent
0   216       0,001199148
1   345       0,001915306
2   580       0,003219934
3   1182      0,006562003
4   2342       0,01300187
5   4654       0,02583719
6   8755       0,04860435
7   15113      0,08390149
8   25147       0,1396063
9   40673       0,2258006
10   62006       0,3442332
11   93798         0,52073
12   139010      0,7717294
13   203963       1,132323
14   327332        1,81722
15   553051       3,070323
16   936168        5,19724
17   1547302      8,590018
18   2421430      13,44284
19   3632266      20,16492
20   4271726      23,71496
21   2934699       16,2923
22   752942        4,18004
23   38092       0,2114719
BitSort end

It's not showing any empty bits but what does it mean when there are bits that have more than 1% of total samples?
9
3rd Party Plugins - (fb2k) / Re: SACD .dsf file conversion plug-ins
Last post by dbnicholls -
Thanks for all of your inputs.  I downloaded and installed FB2k on my Windows 10 computer and installed the SACD add-ins.  After some minimal learning curve, I was able to successfully convert all of my stereo & multichannel DSF files to WAV.  While I haven't checked out the results in detail, the "multichannel" WAV files are significantly larger than the equivalent stereo WAV files, which are significantly larger than the equivalent WAV files of the equivalent MFSL standard CD.

One thing I did note was that FB2k kept telling me that the conversion process was not "lossless", and I did notice a somewhat smaller file size for a converted WAV file than its source DSF file.  I'm guessing that I am not personally going to notice any difference in audio quality.
10
General Audio / Re: Equalise Volumes via dbPoweramp (-db) & wxMp3gain (+db) conversion
Last post by Markuza97 -
Hello!

Only tags are lossless.
All other processes are technically not lossless.
(But they should not affect the quality at all when working with lossless files.)

I am also targeting -14 LUFS myself and I am very happy with it.
-14 LUFS should be equal to 93 dB.

For MP3 files you can simply use foobar2000 to normalize the files directly. This has one disadvantage.
You can only adjust volume in 1.5 dB steps. This is not ideal but it is still better than re-encoding the MP3 files.

If you want to normalize FLAC files you will need to re-convert them into FLAC. Like I said earlier, this is not lossless,
but in reality, there will be no differences, except the volume, of course.

You mentioned WAV files. Why don't you convert them into FLAC to save some space and have proper metadata support?
If you have some huge files you can easily convert everything to 16-bit / 44.1 kHz. Everything above that is really useless.

I have zero experience with MIDI files so I cannot really help you there, sorry.

Edit: I wanted to explain how ReplayGain works so you can understand numbers more easily.

There are two normalization standards.
They are known as "old ReplayGain / ReplayGain Classic" and "new ReplayGain / EBU R128".
Old ReplayGain is targeting 89 dB. New ReplayGain is targeting -18 LUFS.
So 89 dB is equal to -18 LUFS.

By scanning your file, the program will analyze how loud your track is. Let's say that your track is 96 dB / -11 LUFS loud.
Program will then write that your file needs to be adjusted by -7 dB / -7 LUFS.

You can then setup in foobar's playback settings to apply +4 dB / +4 LUFS to target the 93 dB / -14 LUFS.

Think of 89 dB and -18 LUFS as the reference numbers.

Here comes the tricky part. This only applies to specific programs like foobar2000. In other software, the reference point for new EBU R128 is actually -23 LUFS.
So be careful what you are doing.


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