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91
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...  :)
92
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.
93
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?
94
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)
95
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?
96
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.
97
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.


98
General Audio / Equalise Volumes via dbPoweramp (-db) & wxMp3gain (+db) conversion
Last post by hmp -
Hi,
I want to losslessly equalize volumes of my entire music library of mp3s, wavs, FLACs & MIDIs.
Since foobar can't normalize losslessly, I found wxmp3gain can do this to mp3s, but not other audio types.
I discovered that dbpoweramp does it for wavs and FLACs.
FYI: I don't want to simply edit tags for volume equalizing, because I use apps that don't recognize ReplayGain Tags.

I have googled and just simply don't understand the right numbers and how to convert the two db formats to use equalize FLACs in dbpoweramp (via Volume Normalization whilst converting a FLAC to FLAC) to the same level as my mp3s levelled via wxmp3gain.

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.
Once I've chosen a wxmp3gain target (e.g. 75db), since dbpoweramp operates in -db (0 to -40db), whilst wxmp3gain operates in +db (+75db to +105db), how do I convert back and forth from + to - db in dbpoweramp, so that my FLACs and mp3s play at the same volume?

99
Listening Tests / Re: Personal blind sound quality comparison of Opus hard-CBR with framesize options
Last post by C.R.Helmrich -
OK, I figured it out, and explain here so that Jean-Marc (jmvalin) can read it. For stereo, 56 kbit/s CBR uses a different encoder configuration than for 56 kbit/s VBR, one where the encoder adaptively switches between the speech (Silk) and music (CELT) core. 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. Using the --music option in Opus's command-line forces CELT throughout this sample. That avoids the problem you describe on this sample but, of course, a more robust speech/music discriminator would be a better solution.

Chris
100
3rd Party Plugins - (fb2k) / Re: SACD .dsf file conversion plug-ins
Last post by Apesbrain -
Just tested and foobar2000 supports DSF 5.1 conversion to multi-channel WAV, FLAC, AAC and OGG.  I could not find a multi-channel encoder for MP3; Fraunhofer once had one in development, but it's no longer on their site.  I didn't test 7.1 as I have no such files.

After resampling to 44100, I also managed to encode the 5.1 WAV to AC3, if that is useful. Here's the test file:
https://ufile.io/u3uec0xg (Good for 30 days.)
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