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MP3 / Re: best mp3 encoder with something better than a command line interface?
Last post by sven_Bent -
but now, the encoded .mp3 looks like the original .wav. And sounds identical.

Please Follow the TOS especially #8

Atm anything you have said ant be considering of any value much above a drunken mans tale.
You have completely lacked the ability to prove any of your statements  even though you have been asked for several times.

The only thing you have proven so far is a complete lack of understand on any of the tools you have used
- Resampling
- ABX testing
- Spectrograms

Not understading the usage of the tools and not listening to any advice on how to do things correctly is not helping your case.

Even if we regard the clear lack of understanding of tools and lack of evidence. the entire synopsis of your case is outright wrong.
You start by some made up belief that something is wrong and just goes out to say it fixed now. Not even analyzing if your initial issue is present or not.

If you want your case to be taken any kind of serious you need to seriously understand how the basis of proving by evidence works.
All advice I would be able to give you has already been mentioned by others. Try listening to them.

Quote of TOS #8
TOS 8. All members that put forth a statement concerning subjective sound quality, must -- to the best of their ability -- provide objective support for their claims.  Acceptable means of support are double blind listening tests (ABX or ABC/HR) demonstrating that the member can discern a difference perceptually, together with a test sample to allow others to reproduce their findings.  Graphs, non-blind listening tests, waveform difference comparisons, and so on, are TOS not acceptable means of providing support.

For that alone you post should be deleted as the inane rambling it appears to be.
You have been giving plenty of leeway to try to adhere to this, but have rejected any and all of them.
General - (fb2k) / Re: [Suggestions / Wishlists] for future updates
Last post by Sergey77 -
5. Also standardize to displaying of Internet radio links on the playlist for streams: ogg/vorbis; opus; flac
To test (for example) four different internet radio links:
Spoiler (click to show/hide)For the links same behavoir: content for %bitrate%, %samplerate%, %channels%, $meta (genre) (%codec% only for the first link) appears after start playback and disappears after stop playback. The field "General" of section "Properties/Details" is absent constantly.
6. Change behavior of the component "Playback Statistics" for streaming lings items on playlist.
To test (for example) three different links (the first two are HLS):
Spoiler (click to show/hide)first link: The elements of the part “Playback Statistics” in the “Properties/Details” section are missing.
second link: The elements of the part “Playback Statistics” (First/Last played) in the “Properties/Details” are available.
The field "Last modified" of section "Location" in “Properties/Details" is absent constantly for both links.
third link: The field "Last modified" of section "Location" in “Properties/Details" is avaliable.
Content of title field %added% isn't displayed for any internet links at playlist.

Thank you very much for the good changes for Internet radio in v1.4.1 Beta 3. :)
Uploads - (fb2k) / Re: Resampler-V
Last post by jer1956 -
Delete by double clicking the item in the table. I found that out by accident. I deleted something i wanted to keep!
MP3 / Re: best mp3 encoder with something better than a command line interface?
Last post by BrilliantBob -
I feel like I should bin all this "eyes are for listening" nonsense.  It is in no way advancing the original discussion, or even worth having in the first place.
It cannot be helped if BrilliantBob is unwilling to accept the fact that he is misguided.

LAME v3.100 64bit. ATH: using type: 5 (what is it?)

I experimented many types of settings in LAME to obtain a quality .mp3, as transparent as possible related to the original .wav. I disabled Y using VBR. Now the encoding speed is double. I used this batch file for faster bulk encoding (a possible solution too for dnewhous who started this topic)
Code: [Select]
@echo off
color 1E
echo  Ý                                          ÚÄÄÄ¿Þ
echo  Ý         LAME v3.100 64bit unleashed      ³ û ³Þ
echo  Ý                                          ÀÄÄÄÙÞ
echo. ßßßßßßßßßßßßßßßßßßßßßßßßßßßßßßßßßßßßßßßßßßßßßßßßß

:: this batch is very fast for bulk encoding. drag-and-drop the .wav in the batch window or write it with quotes ("FileName.wav")
:: the encoded .mp3 is as transparent as possible related to the .wav source
:: double encoding speed compared with 320 CBR
:: highpass filter disabled. polyphase lowpass filter disabled.
:: Y disabled * VBR 0 enforced to 320 kbps 48,000 Hz 24bit
:: ATH: using type: 5 (what is it?)
:: interchannel masking ratio: 0.0002
:: using LR stereo
::      for access to dev settings (--help dev) put "#define _ALLOW_INTERNAL_OPTIONS 1" in parse.c and compile
:: default lame psychoacustic tuning:                        --ns-bass -0.5 --ns-alto -0.25 --ns-treble -0.025 --ns-sfb21 0.5
:: adjusted psy masking: --ns-bass -8 --ns-alto -8 --ns-treble -8 --ns-sfb21 -16 (psychoacoustic disabled?)

echo ---------------------------
Set /P _infile=drag/enter source:
echo ---------------------------
set _outfile=%_infile:~0,-5%"

lame.exe -ms --ns-bass -8 --ns-alto -8 --ns-treble -8 --ns-sfb21 -16 --interch 0.0002 --verbose -V0 -b256 -B320 -F --resample 48 --bitwidth 24^
 --lowpass -1 --highpass 0.001 --clipdetect %_infile% %_outfile%.mp3 --scale 1

echo ---------------
Set /P _scale=new scale:
echo ---------------
lame.exe -ms --ns-bass -8 --ns-alto -8 --ns-treble -8 --ns-sfb21 -16 --interch 0.0002 --verbose -V0 -b256 -B320 -F --resample 48 --bitwidth 24^
 --lowpass -1 --highpass 0.001 --clipdetect %_infile% %_outfile%.mp3 --scale %_scale%

echo ----------------------------
     Set /P _abort=exit? (y, *):
echo ----------------------------
If /i "%_abort%"=="Y" goto terminate
goto reload

LAME returned these messages and I saw ATH: using type: 5
Code: [Select]
Warning: highpass filter disabled.  highpass frequency too small
LAME 3.100 64bits (
Resampling:  input 44.1 kHz  output 48 kHz
polyphase lowpass filter disabled
Encoding E:\music\09-metallica-of_wolf_and_man.wav
      to E:\music\09-metallica-of_wolf_and_man.mp3
Encoding as 48 kHz stereo MPEG-1 Layer III VBR(q=0)


        scaling: 1
        ch0 (left) scaling: 1
        ch1 (right) scaling: 1
        huffman search: best (outside loop)
        experimental Y=0

stream format:

        MPEG-1 Layer 3
        2 channel - stereo
        padding: all
        variable bitrate - VBR mtrh (default)
        using LAME Tag


        using short blocks: channel coupled
        subblock gain: 1
        adjust masking: -6.8 dB
        adjust masking short: -6.8 dB
        quantization comparison: 9
         ^ comparison short blocks: 9
        noise shaping: 1
         ^ amplification: 2
         ^ stopping: 1
        ATH: using
         ^ type: 5
         ^ shape: 1 (only for type 4)
         ^ level adjustement: -7.1 dB
         ^ adjust type: 3
         ^ adjust sensitivity power: 1.000000
        experimental psy tunings by Naoki Shibata
           adjust masking bass=-8.5 dB, alto=-8.25 dB, treble=-8.025 dB, sfb21=-
15.5 dB
        using temporal masking effect: no
        interchannel masking ratio: 0.0002

    Frame          |  CPU time/estim | REAL time/estim | play/CPU |    ETA
 10695/10695 (100%)|    0:43/    0:43|    0:44/    0:44|   5.8555x|    0:00
256 [   20] %
320 [10675] %%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%
   kbps        LR  %     long switch short %
  319.9      100.0        93.7   3.6   2.6
Writing LAME Tag...done
ReplayGain: -5.9dB
WARNING: clipping occurs at the current gain. Set your decoder to decrease
         the  gain  by  at least 0.7dB or encode again using  --scale 0.92
         or less (the value under --scale is approximate).

new scale:

Never heard before about ATH type 5. I don't post spectrograms to not make people angry, but now, the encoded .mp3 looks like the original .wav. And sounds identical.
Opus / Re: Opus 1.3 is out!
Last post by lvqcl -
Actually, that's one of the few things compilers are *not* allowed to do when optimizing.
You mean, to replace (a+b)+c with a+(b+c)?  Well, Visual Studio Opus project has FloatingPointModel==Fast in Release configuration.

According to MSDN,
The compiler may reorder operations or perform algebraic transforms—for example, by following associative and distributive rules—without regard to the effect on finite precision results.
General - (fb2k) / Re: Skip Silence DSP accidentally enabled for months while ripping music
Last post by Case -
What does this -90db mean anyway? Does it take 100db as default value and anything below 10db gets cut away? Really wonder.
The reference level is digital fullscale. -90 dB means anything with absolute sample value of about 0.0000316 (on a scale from 0...+1) is considered silence. With 16 bit material the lowest sample value that isn't silence is -90.31 dBFS.
Opus / Re: Opus gapless and glitchness encoding
Last post by j7n -
The original implementation for replaygain in opus would not cause clicks as the album gain would be used by default. It did cause playback to be 5 dB lower outside of Foobar going against the ca. -18 dB RG standard established for over 10 years, and even earlier as it approximately matches levels on old CDs too. The header gain isn't even supposed to be at R.128, and could be chosen freely. But that is in the past as now R128_ALBUM_GAIN can be used, at the expense of compatibility with players that don't support it, which the original method gave for free – but at wrong reference level.

Opus is quite a bit worse than other codecs in gaplessness with flawed source material with low frequency osscillations or DC offset. Those are errors that should be fixed at the source, if they are noticed. But it would be nice if a fully gapless Opus encoder (in Foobar or similar GUI program) could deal with this and the same time improve the samples by filtering the DC.
Opus / Re: Opus 1.3 is out!
Last post by jmvalin -
Different optimizations -> different order of floating-point instructions -> different results.
For example, (a+b)+c usually is not the same as a+(b+c).

Actually, that's one of the few things compilers are *not* allowed to do when optimizing. OTOH, the compiler *is* allowed to use more bits than required by the program (especially on old x87 FPUs that had 80-bit float, but I think even for 32->64 bit float). Also, Opus uses transcendental math functions (sin, exp, log, ...) that are not required by the relevant standards to be accurate down to the LSB rounding. I suspect there may be a few other details that can cause differences, but the order of operations definitely must be preserved, otherwise overflows and underflows could cause massively wrong results.
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