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1
I think you're confusing bit-RATE with bit-DEPTH.
mp3 has no bit-depth. When decoded and played, they'll probably all be 16-bit, according to your sound card settings, so you don't have to think about that when compressing to mp3.

16kbps is the bitrate, the amount of data per second, and it's very low. Try something higher. 64 is also low, but usually okay for speech. You're also transcoding from lossy to even more lossy, wich will enlarge any artifacts that were in the 64kbps version.

You might be able to get away with a lowpass of 8KHz or lower, but you'll have to test a few files and see if it sounds okay.
2
... Of course only EQ without phase correction always makes sound worse.
I disagree with this statement.

The magnitude errors that we want to correct are usually not linear-phase. When applying a (usually minimum-phase) equalizer to flatten the magnitude, the change in phase response is hard to predict without measurements.

"Worse sound" implies listening tests, where it is usually found that moderate phase errors are quite inaudible.
 
-k
3
There is an outline of the  two-step process to avoid overcomplicated scripting
https://www.youtube.com/watch?v=YFcw6H8ht1A
Debate somewhere on this very forums (can't find right now).
4
Don't get me wrong. What I was talking about are not specific to qobuz, UMG or any company, just a remind that don't automatically assume things will sound the same just because the medium or format itself is capable of transparency.

Also, if you look at Wombat's post:
https://hydrogenaud.io/index.php/topic,111198.0.html

Apart from the suspected watermark from 1-5kHz, there are also differences from 14-22kHz. I am not going to talk about the audibility of such differences because I don't have those songs, but such differences are totally avoidable with today's resampling technology.

Okay, now I got you and how your post was intended. For the High Frequency differences I must honestly say: I don't care. In my ears, these are not the frequencies where listening-joy or -pain comes from. If the sound lacks some High End, I just raise the 8 and 16 kHz EQ bands of my player a bit, or lower them. I'm fine with that :)

Another thing is, of course the watermark. I tried pull every EQ band down, except 1, 2 and 4 kHz, hoping that I could identify a watermark in my qobuz purchased albums. I'm not sure if this could help identifying a watermark. I thought it could, since less frequencies are covering the mid-range, where the watermark is expected. However, I was not able to hear anything suspicious in my case. Still I wanna check, what the difference is between my version, and the one which is sold on CD and could pass AR successfully. I know someone who I can borrow the CD version of, and I will rip that disc and see the diff between the qobuz and the CD version. Looking forward to share the result with you.
5
General Audio / Re: M3U playlist support (Foobar2K) ?
Last post by Case -
They work just fine. The extra metadata in the playlists is just ignored. foobar2000 displays the metadata it is configured to display that it reads from the files directly.
6
Hi I am Rupesh from India. I have a huge directory of size 93.5 gb with 8500 mp3 files and 2000 sub directories.

All these mp3s are speeches recorded by someone at 64 kbps. I want to compress these files recursively using lame with 16 kbps bit rate and 11050 sample rate.

I have found some guis or front end for lame like lamexp,razorlame,lamefrontend none has options to compress mp3 files recursively at 16 kbps. On searching net I found a script for ffmpeg to compress recursively. I have deleted ffmpeg code and substituted lame code instead of which I am providing below

Code: [Select]
.
@echo off
setlocal EnableExtensions DisableDelayedExpansion

rem // Define constants here:
set "_SOURCE=I:\to convert3"
set "_TARGET=H:\converted\lame4"

rem // Change to source directory temporarily:
pushd "%_SOURCE%" || exit /B 1
rem // Enumerate source files, return paths relative to the source directory:
for /F "delims=" %%F in ('xcopy /L /S /I ".\*.mp3" "%_TARGET%" ^| find ".\"') do (
    echo Currently converting "%%F"...
    rem // Create destination directory, suppress error if it aready exists:
    mkdir "%_TARGET%\%%F\.." 2> nul
    rem // Perform actual file conversion, using paths relative to target directory:
    lame    --abr 16 -m j -q 0 --resample 22.05 --priority 4   "%%F" "%_TARGET%\%%F\..\%%~nF.mp3"
)
echo Completed.
popd

endlocal

The above script runs only on Windows but can't run on Linux and it is better to run the script on Linux.

Please try to convert the above Windows batch script to Linux shell script and post it.

Please try to suggest if you know a front end for lame which can compress mp3 files recursively at 16 kbps.

Regards,
Rupesh.
7
Hi I am Rupesh from India. I have a huge directory of size 93.5 gb with 8500 mp3 files and 2000 sub directories.

All these mp3s are speeches recorded by someone at 64 kbps. I want to compress these files recursively using lame with 16 kbps bit rate and 11050 sample rate.

I have compressed the above huge directory with above options using ffmpeg and the resulted destination directory size was 29.5 gb and the quality was not good. Then I thought instead of compressing using ffmpeg it is better to compress using lame.

I have compressed some directories with options given below using lame and the quality was not good.

Code: [Select]
 lame    --abr 20 -m j -q 0 --resample 22.05 --priority 4
lame    --abr 16 -m j -q 0 --resample 11.025 --priority 4
lame    --abr 16 -m j -q 0 --resample 22.05 --priority 4

The files I converted using ffmpeg was better than the files converted using lame.

In the above sound quality was not good I mean after the compression completed when I try to play these files on any music player I am getting    sounds like sshhh... garrr... and some electronic noises.

Please try to suggest optimal settings to compress these speech mp3 files I mean less noises like sshhh garrrr and other such electronic noises.
Are there any other ways to reduce these noises during the process of compression.

I think that sample rate and bit rate play a lead role together in compression I mean it is not feasible to use 44.1 kHz to 16 kbps bit rate and so suggest the correct sample rate to 16 kbps and 24 kbps.

If it is not possible to eliminate these noises in 16 kbps bit rate try to suggest at other bit rates upto 24 kbps.

As I am new to audio compression please suggest optimal settings I mean bitrate below 24 kbps, sample rate to the corresponding bitrate, replay gain, athtype, highpass, low pass etc., found in lame documentation.


Regards,
Rupesh.
8
Support - (fb2k) / Re: Query search retrieving bad results.
Last post by Case -
THAN is indeed incorrect. It's not a keyword and gets treated as text. Chances are your query returned every file in your library.
9
Well okay, I hear what you mean, but is it really the case that the CD version differs from the ones available in the download stores? I just had a quick listen into the previews of that album on qobuz, and (at least on cheap office headphones) it sounded also pretty distorted, just like the CD version against the GH version. Furthermore, after reading about this, it's no case of crippled mastering here, because it seems that the mastering engineer already got the tracks with that killer distortion. It's just logical that there is a differnce between a GH version of a song and the sold version - the GH version must be produced differently to focus the guitars for that game. Anyway - for me this is no proof that music will be mastered differently for download stores compared to CDs (but I will again listen to the qobuz previews of Death Magnetic later on...)
Don't get me wrong. What I was talking about are not specific to qobuz, UMG or any company, just a remind that don't automatically assume things will sound the same just because the medium or format itself is capable of transparency.

Also, if you look at Wombat's post:
https://hydrogenaud.io/index.php/topic,111198.0.html

Apart from the suspected watermark from 1-5kHz, there are also differences from 14-22kHz. I am not going to talk about the audibility of such differences because I don't have those songs, but such differences are totally avoidable with today's resampling technology.
10
General Audio / Re: Sound "drop-outs" every ten seconds or so
Last post by Juha -
If you want to look closer what happens in your system at that point you get the drop-out then just use perfmon bundled with OS.
Just use suitable counters there.