Posts by igorzwx

igorzwx · Freedom Hacks

Yes, it was not a scientific experiment, and it was not intended.
The fft resampler was tested several years. Then it was replaced with dct in an OSS4 plugin.
One user noticed the difference and began to complain about the sound quality and some unwanted effects, such as "fatigue".

igorzwx · Freedom Hacks

Yes, a blind test might be very useful, but the problem is that you may not know what you are actually testing.

If, for example. a sound system performs resampling and format conversions in software, you may not know what you are actually playing with your DAC, or soundcard. That is why, perhaps, some users want to know how to enable "exclusive mode".

steve_v wrote:

Trivial differences in resampling are irrelevant, and unless you are doing digital mastering, so are bit depths >16 and sampling rates significantly above the Nyquist limit. Fight me.

Although the difference between Petrov's dct and fft seems insignificant, I would strongly recommend to use fft instead of dct (default). The reason is the results of an unintended blind test with OSS4 users.

igorzwx · Freedom Hacks

Again, my hypothesis is that the Playback Designs MPD-8 DAC is playing 32bit float DXDs (which it should not play), because a sound system, or a player "performs the necessary format conversions in software".

Would you agree that it might be the case?

igorzwx · Freedom Hacks

aluma wrote:

Can you honestly say why you are raising these questions?

I asked the question about the Playback Designs MPD-8 DAC, because you seems to be an expert on DACs.

The Playback Designs MPD-8 accepts only 24-bit input files, so how can this be?
_https://positive-feedback.com/reviews/music-reviews/what-we-hear-with-dxd-32-bit-files/

Instead of answering the question, you are trying to sell me a very old theory that people do not need high resolution audio, because they cannot hear the difference.

I've talked with the Playback Design's Andreas Koch about this and he has a theory but no firm answer.
_https://positive-feedback.com/reviews/music-reviews/what-we-hear-with-dxd-32-bit-files/

It seems that you also have theories but no firm answer.

Perhaps, you may try to help us to find out how it can be that the Playback Designs MPD-8 DAC is playing 32bit float DXDs, which it should not play.

With OSS4, one may suspect the COOKEDMODE:

$ cat /usr/lib/oss/conf/osscore.conf | grep sampling -A9
# By default OSS will let applications use sampling rates and formats that
# are not supported by the hardware. Instead OSS performs the necessary
# format conversions in software. Applications that don't tolerate these
# kind of conversions usually disable them by using features of the OSS API
# (SNDCTL_DSP_COOKEDMODE). If this option is set to 0 then the format
# conversions will be disabled for all applications and devices unless the
# application explicitly enables them. This option should not be changed
# without wery strong reason.
# 
#cooked_enable=1

What do you think about such hypothesis?

igorzwx · Freedom Hacks

Yes, it is very interesting.

But we still have a question to answer:

But wait, you say... The Playback Designs MPD-8 accepts only 24-bit input files, so how can this be?
_https://positive-feedback.com/reviews/music-reviews/what-we-hear-with-dxd-32-bit-files/

The DXD wave is 32bit Float. What do you think about this?

igorzwx · Freedom Hacks

If you cannot hear the difference between Petrov's dct and fft resamplers, you may try to measure it with exact scientific instruments.

The method is very simple:
1. downsample a DXD wave,
2. upsample the result to DXD format,
3. measure the difference with the original DXD wave,
4. compare the results of measurement for different resamplers.

Perhaps, you noticed that Petrov's pcm-utils package provides a sort of toolbox:
- pcm_info: information of the data format in 'wav' file
- pcm_conv: standalone PCM converter
- pcm_mse: computing of the Mean Squared Error (MSE) between two 'wav' files in PCM format

Free DXD samples are available here:
What We Hear With DXD 32-bit Files (Free Sample Downloads)
_https://positive-feedback.com/reviews/music-reviews/what-we-hear-with-dxd-32-bit-files/

[FFT] Downsampling:

$ pcm_conv -f 192000 -b 32 -T fft -v *32bit.wav Faust_192kHz_32bit_downfft.wav
Input file : 08-Faust - Funeral March Of A Marionette - 32bit.wav
 Samples rate    = 352800 Hz
 Channels        = 2
 Bits per sample = 32 (actual: 32, float)

Output file: Faust_192kHz_32bit_downfft.wav
 Samples rate    = 192000 Hz
 Channels        = 2
 Bits per sample = 32 (actual: 32, fixed)

        FL      FR      
FL      1.000   -----   
FR      -----   1.000   

Windows : "Vorbis" (37632 => 20480)
Delay   : 107 ms
Sync.   : ON
Trans.  : "FFT"

[|||||||||||||||||||||||||||||||||||||||||||||||||||]  99.0 %
Ok.

File duration  : 275.906667 sec
Processing time: 13.305000 sec ( 4.82% of real time)

[FFT] Upsampling:

$ pcm_conv -f 352800 -b 32f -T fft -v *downfft.wav Faust_DXD_32bit_downupfft.wav
Input file : Faust_192kHz_32bit_downfft.wav
 Samples rate    = 192000 Hz
 Channels        = 2
 Bits per sample = 32 (actual: 32, fixed)

Output file: Faust_DXD_32bit_downupfft.wav
 Samples rate    = 352800 Hz
 Channels        = 2
 Bits per sample = 32 (actual: 32, float)

        FL      FR      
FL      1.000   -----   
FR      -----   1.000   

Windows : "Vorbis" (20480 => 37632)
Delay   : 107 ms
Sync.   : ON
Trans.  : "FFT"

[|||||||||||||||||||||||||||||||||||||||||||||||||||]  99.0 %
Ok.

File duration  : 275.786667 sec
Processing time: 12.539000 sec ( 4.55% of real time)

[FFT] Measuring the difference:

$ pcm_mse *32bit.wav *upfft.wav -b 10 -e 10000000000
Opening file '08-Faust - Funeral March Of A Marionette - 32bit.wav' ... Ok.
 Samples rate    = 352800 Hz
 Channels        = 2
 Bits per sample = 32 (actual: 32, float)

Opening file 'Faust_DXD_32bit_downupfft.wav' ... Ok.
 Samples rate    = 352800 Hz
 Channels        = 2
 Bits per sample = 32 (actual: 32, float)

Processing format:
 Samples rate    = 352800 Hz
 Channels        = 2
 Bits per sample = 64 (actual: 64, float)

File '08-Faust - Funeral March Of A Marionette - 32bit.wav' is longer than 'Faust_DXD_32bit_downupfft.wav'

[|||||||||||||||||||||||||||||||||||||||||||||||||||]  99.0 %
Ok.

Fragment form 10 to 97259904 (97259894 samples)

Front Left:
~~~~~~~~~~~
 Dynamic range: from -0.92281 to 0.792051
 MSE = 0.0001395137
 SNR = 38.5538 dB
 First diff.: 0.000288079 on position 10 (0 in fragment)
 Max diff.  : 0.00377898 on position 81946564 (81946554 in fragment)
 Number of different samples: 97259747 ( 100.0% )

Front Right:
~~~~~~~~~~~~
 Dynamic range: from -0.984298 to 1.04113
 MSE = 0.0001161680
 SNR = 39.3491 dB
 First diff.: 0.000536871 on position 10 (0 in fragment)
 Max diff.  : 0.0408844 on position 66347702 (66347692 in fragment)
 Number of different samples: 97259754 ( 100.0% )

[DCT] Downsampling:

$ pcm_conv -f 192000 -b 32 -T dct -v *32bit.wav Faust_192kHz_32bit_downdct.wav
Input file : 08-Faust - Funeral March Of A Marionette - 32bit.wav
 Samples rate    = 352800 Hz
 Channels        = 2
 Bits per sample = 32 (actual: 32, float)

Output file: Faust_192kHz_32bit_downdct.wav
 Samples rate    = 192000 Hz
 Channels        = 2
 Bits per sample = 32 (actual: 32, fixed)

        FL      FR      
FL      1.000   -----   
FR      -----   1.000   

Windows : "Vorbis" (37632 => 20480)
Delay   : 53 ms
Sync.   : ON
Trans.  : "DCT"

[|||||||||||||||||||||||||||||||||||||||||||||||||||]  99.0 %
Ok.

File duration  : 275.906667 sec
Processing time: 7.655000 sec ( 2.77% of real time)

[DCT] Upsampling:

$ pcm_conv -f 352800 -b 32f -T dct -v *downdct.wav Faust_DXD_32bit_downupdct.wav
Input file : Faust_192kHz_32bit_downdct.wav
 Samples rate    = 192000 Hz
 Channels        = 2
 Bits per sample = 32 (actual: 32, fixed)

Output file: Faust_DXD_32bit_downupdct.wav
 Samples rate    = 352800 Hz
 Channels        = 2
 Bits per sample = 32 (actual: 32, float)

        FL      FR      
FL      1.000   -----   
FR      -----   1.000   

Windows : "Vorbis" (20480 => 37632)
Delay   : 53 ms
Sync.   : ON
Trans.  : "DCT"

[|||||||||||||||||||||||||||||||||||||||||||||||||||]  99.0 %
Ok.

File duration  : 275.840000 sec
Processing time: 7.881000 sec ( 2.86% of real time)

[DCT] Measuring the difference:

$ pcm_mse *32bit.wav *updct.wav -b 10 -e 10000000000
Opening file '08-Faust - Funeral March Of A Marionette - 32bit.wav' ... Ok.
 Samples rate    = 352800 Hz
 Channels        = 2
 Bits per sample = 32 (actual: 32, float)

Opening file 'Faust_DXD_32bit_downupdct.wav' ... Ok.
 Samples rate    = 352800 Hz
 Channels        = 2
 Bits per sample = 32 (actual: 32, float)

Processing format:
 Samples rate    = 352800 Hz
 Channels        = 2
 Bits per sample = 64 (actual: 64, float)

File '08-Faust - Funeral March Of A Marionette - 32bit.wav' is longer than 'Faust_DXD_32bit_downupdct.wav'

[|||||||||||||||||||||||||||||||||||||||||||||||||||]  99.0 %
Ok.

Fragment form 10 to 97297536 (97297526 samples)

Front Left:
~~~~~~~~~~~
 Dynamic range: from -0.92281 to 0.792051
 MSE = 0.0001395591
 SNR = 38.5524 dB
 First diff.: 0.000289127 on position 10 (0 in fragment)
 Max diff.  : 0.0037787 on position 81946564 (81946554 in fragment)
 Number of different samples: 97297377 ( 100.0% )

Front Right:
~~~~~~~~~~~~
 Dynamic range: from -0.984298 to 1.04113
 MSE = 0.0001162062
 SNR = 39.3477 dB
 First diff.: 0.000540629 on position 10 (0 in fragment)
 Max diff.  : 0.0409828 on position 66347702 (66347692 in fragment)
 Number of different samples: 97297363 ( 100.0% )

Results of measurement:

[FFT] Front Left:    MSE = 0.0001395137
[DCT] Front Left:    MSE = 0.0001395591

[FFT] Front Right:   MSE = 0.0001161680
[DCT] Front Right:   MSE = 0.0001162062

MSE is the Mean Squared Error 

[FFT downsampling] Processing time: 13.305000 sec ( 4.82% of real time)
[DCT downsampling] Processing time: 7.655000 sec ( 2.77% of real time)

You can see that DCT is faster than FFT (13.305/7.655 ≈ 1.738).
FFT is more exact, although the difference is insignificant.
Petrov claimed that the difference is not audible, but it was detected by an OSS4/Gentoo user in a "blind test".

If you are using ALSA, you may not hear the difference between 192kHz/32bit and 48kHz/16bit formats of the same audio file (if the DXD wave was downsampled with the FFT resampler).

You may also try to measure the Secret Rabbit Code of ALSA (libsamplerate)
_https://libsndfile.github.io/libsamplerate/
_https://github.com/libsndfile/libsamplerate

sudo apt install sndfile-programs

man sndfile-convert

igorzwx · Freedom Hacks

aluma wrote:

An ordinary computer is not suitable as a playback source.

I can agree with this. The same is for recording. A sort of portable HiRes audio recorders, such as Olympus or Zoom, might be a much more reasonable solution.

But, it we take this for granted, then, perhaps, Petrov's fftrate ALSA plugin with Audacious (without pulseaudio or pipewire) might be a practical solutions for notebooks.

aluma wrote:

Today I heard wonderful sound “out of the box” only from my son’s Mac, both desktop and laptop.

This is exactly the case. Kids are perfectly happy with Mac. Then they want to have the same sound quality with Fedora on a cheap notebook. It seems that fftrate plugin and apulse fixed the problem.

Notice, however, that MacOS sound system does not support 192kHz sample rate. But you can downsample DXD waves to 96kHz with Petrov's pcm_conv. Then you can play them with foobar2000. It can be installed with Homebrew
_https://formulae.brew.sh/cask/foobar2000#default

Foobar2000 "exclusive mode" is now supported for MAC. It may work, if the audio device supports the bit rate and sample rate of your audio file. There might be other problems as well
_https://forum.rme-audio.de/viewtopic.php?id=37063

_https://support.apple.com/en-gb/108326
Play high sample rate audio on your Mac
The hardware digital-to-analogue converter (DAC) built into compatible Mac computers supports sample rates of up to 96 kHz.
Compatible Mac computers feature a high-quality, built-in hardware DAC that can convert up to 96 kHz digital audio to analogue audio. You can connect analogue devices such as headphones or speakers directly to the headphone jack on your Mac and monitor your audio at full resolution without needing an external DAC.
To set the sample rate for the headphone jack, use the Audio Midi Setup app, which is located in the Utilities folder of your Applications folder.
_https://support.apple.com/en-gb/guide/audio-midi-setup/ams59f301fda/mac

ArchWiki wrote:

By default the sample rate is 48000hz. There are several conditions in which you may want to change this. This all depends on your usage patterns. You want the sample rate you are using to match the media you use the most. If your computer has to change the sampling rate of the media to suit the hardware it is likely, though not guaranteed, that you will have a loss in audio quality. This is most noticeable in down sampling (ie. 96000hz → 48000hz). There is an article about this issue in Stereophile which was discussed on Apple's CoreAudio API mailing list if you wish to learn more about this issue.
_https://wiki.archlinux.org/title/Open_Sound_System#Changing_the_sample_rate
_https://www.stereophile.com/news/121707lucky/
_https://lists.apple.com/archives/coreaudio-api/2008/Jan/msg00272.html

This was a quote from ArchWiki manual on OSS4. All this does not apply to Petrov's fftrate resampler. Regardless of your usage patterns, you can safely configure fftrate for the highest sample rate supported by your sound card. Some users even claimed that Petrov's fftrate resampler somehow improves sound quality of audio files. I cannot dismiss their claims, because one of them detected the difference between Petrov's dct and fft resamplers in a "blind test" (he noticed the difference and began to complain). However, you have to dismiss such claims, if you know what the Gibbs phenomenon is. The problem is knowledge, of course. It always restricts our freedom of thinking.

In a word, you can easily switch between resamplers in /etc/fftrate.conf

$ cat /etc/fftrate.conf | grep Transform -A3
# Transform type
# Available: dct, fft (default: dct)
#transform = dct
transform = fft

igorzwx · Freedom Hacks

The problem seems be that ALSA does not permit "exclusive mode", and, therefore, it may not make much sense to use a very expensive DAC with ALSA.

igorzwx · Freedom Hacks

Thank you for the information. Have you tried it?

igorzwx · Freedom Hacks

Perhaps, you are missing the point. iMic USB sound card, or a similar one, can be used for video conferencing or playing videos in youtube, if fftrate plugin and apulse are installed, and PulseAudio and PipeWire are removed. This might be a practical solution for notebooks.
Intel HDA codecs (32bit 192kHz) can be used with OSS4 for playing HiRes audio files in "exclusive mode". Although, of course, Intel HDA codecs can also be used with ALSA, if you cannot compile OSS4.

However, if you suffer from hearing loss caused by PulseAudio, you may need an amplifier.

If you want to explain us how to play DXD and DSF audio files on Devuan with the highest possible sound quality, you may start a new topic.

igorzwx · Freedom Hacks

Free DXD samples are available here:
What We Hear With DXD 32-bit Files (Free Sample Downloads)
_https://positive-feedback.com/reviews/music-reviews/what-we-hear-with-dxd-32-bit-files/

If you do not know how to play DXD waves and flacs (and dsf files) on Devuan, it may mean, perhaps, that your freedom is somehow restricted by your knowledge. They can be easily played with Audacious through the help of the fftrate resampler.

The DXD samples were provided as a sort of "material artefacts" for a theological discussion about 32bit format:

if PCM processing has been applied in post, or if the recording is originally a PCM recording (viz. DXD), my experience has been that the 32-bit version simply sounds better than the either the 24-bit file or the PCM processed DSD256 file. It's a half-step closer to the sound I love from a Pure DSD256 recording.
At least on my primary system, with my DAC (a Playback Designs MPD-8).
But wait, you say... The Playback Designs MPD-8 accepts only 24-bit input files, so how can this be?
I don't know the answer. I've talked with the Playback Design's Andreas Koch about this and he has a theory but no firm answer.
_https://positive-feedback.com/reviews/music-reviews/what-we-hear-with-dxd-32-bit-files/

See also: "What Is 32-Bit Float Audio, and Should You Record In It?"
_https://www.wired.com/story/32-bit-float-audio-explained/

Since Intel HDA codecs support 32bit format, you can play 32bit audio files with OSS4 or ALSA, without thinking "how can this be?" Although, of course, Intel HDA codecs do not support "32bit Float" audio format.

1. DXD and vinyl

If you know what the Gibbs phenomenon is, you may think that analogue recordings might be better than digital ones.

A phonograph record (also known as a gramophone record, especially in British English), a vinyl record (for later varieties only), or simply a record or vinyl is an analog sound storage medium in the form of a flat disc with an inscribed, modulated spiral groove.
_https://en.wikipedia.org/wiki/Phonograph_record

However, when sample rate 🡪 infinity, digital recordings may eventually converge to analogue ones.
The goal of the "never-ending quest for the ultimate sound quality" seems to be a sort of extreme freedom.

2. Digital eXtreme Definition (DXD) with OSS4

There is a Petrov's fftrate plugin for Audacious, but it works only with a very old version of Audacious. Therefore, if you want to play DXD waves (or dsf files) with OSS4 in "exclusive mode", you may better convert them to an audio format, which is supported by your sound card.

For example, Intel HDA codecs support 32bit 192kHz format:

$ ossinfo -v9 | grep "Output formats" -A8
    Output formats (0x00001010):
      AFMT_S16_LE	- 16 bit signed little endian
      AFMT_S32_LE	- 32 bit signed little endian
    Device handle: PCIa0021458-0000:00:1b.0-au01
    Related mixer dev: 0
    Sample rate source: 0
    Preferred channel configuration: Not indicated
    Supported number of channels (min - max): 2 - 8
    Native sample rates (min - max): 44100 - 192000 (44100,48000,96000,192000)

DXD waves can be converted to 192kHz 32bit format with Petrov's pcm_conv. Notice that the original DXD wave is 32bit float:

$ pcm_conv -f 192000 -b 32 -T fft -v "08-Faust - Funeral March Of A Marionette - 32bit.wav" output_Faust_192kHz_32bit.wav
Input file : 08-Faust - Funeral March Of A Marionette - 32bit.wav
 Samples rate    = 352800 Hz
 Channels        = 2
 Bits per sample = 32 (actual: 32, float)

Output file: output_Faust_192kHz_32bit.wav
 Samples rate    = 192000 Hz
 Channels        = 2
 Bits per sample = 32 (actual: 32, fixed)

        FL      FR      
FL      1.000   -----   
FR      -----   1.000   

Windows : "Vorbis" (37632 => 20480)
Delay   : 107 ms
Sync.   : ON
Trans.  : "FFT"

[|||||||||||||||||||||||||||||||||||||||||||||||||||]  99.0 %
Ok.

File duration  : 275.906667 sec
Processing time: 14.542000 sec ( 5.27% of real time)

Now you can play it with OSS4 in "exclusive mode":

ossplay -R -d0 -s192000 -c2 -fS32_LE output_Faust_192kHz_32bit.wav

You can also play 192kHz waves with Audacious in "exclusive mode", if OSS4 plugin is enabled.

dsf files can be easily converted to waves with Audacious:
File → Settings → Audio
(1) set "Bit depth" to "Floating point"
(2) "Output plugin": "FileWriter Plugin" (then "Settings")
(3) "FileWriter Plugin Settings" - Output file format: WAV
Now, you can simply "play" a dsf file with Audacious, and it will be converted to a wave.
Notice, however, that the wave might be very big:

$ mediainfo '08-Faust - Funeral March Of A Marionette - DSD256.dsf.wav'
General
Complete name                            : 08-Faust - Funeral March Of A Marionette - DSD256.dsf-1.wav
Format                                   : Wave
Format settings                          : PcmWaveformat
File size                                : 2.90 GiB
Duration                                 : 4 min 35 s
Overall bit rate mode                    : Constant
Overall bit rate                         : 90.3 Mb/s

Audio
Format                                   : PCM
Format profile                           : Float
Codec ID                                 : 3
Codec ID/Hint                            : IEEE 
Duration                                 : 4 min 35 s
Bit rate mode                            : Constant
Bit rate                                 : 90.3 Mb/s
Channel(s)                               : 2 channels
Sampling rate                            : 1 411 kHz
Bit depth                                : 32 bits
Stream size                              : 2.90 GiB (100%)

It can be converted to 32bit 192kHz with pcm_conv:

$ pcm_conv -f 192000 -b 32 -T fft -v *DSD256.dsf.wav output_Faust_192kHz_32bit.dsf.wav
Input file : 08-Faust - Funeral March Of A Marionette - DSD256.dsf.wav
 Samples rate    = 1411200 Hz
 Channels        = 2
 Bits per sample = 32 (actual: 32, float)

Output file: output_Faust_192kHz_32bit.dsf.wav
 Samples rate    = 192000 Hz
 Channels        = 2
 Bits per sample = 32 (actual: 32, fixed)

        FL      FR      
FL      1.000   -----   
FR      -----   1.000   

Windows : "Vorbis" (150528 => 20480)
Delay   : 107 ms
Sync.   : ON
Trans.  : "FFT"

[|||||||||||||||||||||||||||||||||||||||||||||||||||]  99.0 %
Ok.

File duration  : 275.906667 sec
Processing time: 61.711000 sec (22.37% of real time)

Now, it can be played in "exclusive mode":

ossplay -R -d0 -s192000 -c2 -fS32_LE output_Faust_192kHz_32bit.dsf.wav

3. DXD and dsf with ALSA

DXD waves and flacs (as well as dsf files) can be easily played with Audacious, if Petrov's fftrate ALSA plugin is installed (and configured).

It is also possible to output a sort of "debug" to terminal or a text file. This is how it looks with iMic USB sound card (16bit 48kHz):

$ audacious 2>&1 *.dsf
Input:  1411200 Hz, 2 ch, 's32_le' (0xa): dummy = 0, period = 56448
Output: 48000 Hz, 2 ch, 's16_le' (0x2): dummy = 0, period = 1920
Rates:  1411200 --> 48000 (J: 0.00%, T: FFT, W: Vorbis)
Ok.

$ audacious 2>&1 *32bit.wav
Input:  352800 Hz, 2 ch, 's32_le' (0xa): dummy = 0, period = 14112
Output: 48000 Hz, 2 ch, 's16_le' (0x2): dummy = 0, period = 1920
Rates:  352800 --> 48000 (J: 0.00%, T: FFT, W: Vorbis)
Ok.

You may also try FLAC (encoding and decoding) with "32bit fixed" audio format.

FLAC 1.4.0 released
09 Sep 2022
This release brings many small improvements and a few large ones. Here are the most important for end-users:
FLAC can now encode and decode 32 bit-per-sample audio. This means the FLAC format is now one step closer to being fully implemented. Note that this is 32 bit integer samples, not 32 bit float samples.
_https://xiph.org/flac/2022/09/09/flac-1-4-0-released.html

igorzwx · Off-topic

If reading does not help, you may try wu wei.

igorzwx · Off-topic

It might be a sin to obfuscate the Teaching.

Our life is shaped by our mind; we become what we think.
The Dhammapada - Full Audiobook with Text (AudioEbook)
_https://youtu.be/TcGfY4WSgVA

igorzwx · Off-topic

Yes, I was confused by the movie, so many facts...

igorzwx · Off-topic

The problems of "vital resources" and of "reproduction" are real problems. Another problem is how to separate facts from myths.

You might be surprised, but facts are myths, and nothing more. This scientific discovery was made by Roland Barthes.
If you think that it is a joke, you might be mistaken. The secret knowledge in encoded in British textbooks:

Barthes claims that is at the level of secondary signification or connotation that myth is produced and consumed... According to Barthes,
"what allows the reader to consume myth innocently is that he does not see it as a semiological system but as an inductive one. Where there is only equivalence, he sees a kind of causal process: the signifier and the signified have, in his eyes, a natural relationship. This confusion can be expressed otherwise: any semiological system is a system of values; now the myth-consumer takes the signification for a system of facts: myth is read as a factual system, whereas it is but a semiological system."
Jonh Storey, And Introductory Guide to Cultural Theory and Popular Culture, Edinburgh University Press, 1993.

To summarize, facts are myths, and myths are a sort of knowledge. The consumer of facts is a slave of knowledge.

igorzwx · Off-topic

Yes, it is very interesting.

igorzwx · Off-topic

Sorry to interrupt the voting procedure.

rolfie wrote:

Didn't get audio to work with pure ALSA.

This seems to be the problem.

There is a belief that problems can be solved by voting. If it does not work, we may try something else.

igorzwx · Off-topic

Most important, it is easy to remove. Just another crap produced by the same sort of people.

I was asked to fix sound quality on Fedora without removing PipeWire.
Sound quality was so bad that it was impossible to use Discord.
First of all, I installed fftrate plugin and configured it for 32bit 192kHz.
The user was already satisfied with sound quality.
Then, I installed apulse and explained how to use it. This produced a sort of happiness.

igorzwx · Off-topic

Can you hear ordinary music recordings without amplification?
If you feel that you really need an amplifier, you may better visit a doctor. Otherwise, very soon, you may need a much more powerful amplifier.

igorzwx · Off-topic

Your sound card can reproduce the sound, because it has a very powerful amplifier (as you proudly declared before).
I may agree that such sound cards might be very helpful for semi-deaf PulseAudio users.

igorzwx · Off-topic

It has nothing to do with my sound card. The audio file was amplified with Audacity.
The strong and unpleasant noise can be easily detected on spectrograms.
You can install Audacity (sudo apt install audacity) and verify this.

Perhaps, you do not know how Audacity works. You open an audio file with Audacity. Select the whole file and amplify it. Save the result. Then you can select a time interval and build a spectrogram.

If you think that Audacity's amplifier (with default settings) is producing strong audio distortions, you can file a bug.

> mediainfo 2bit64_fragment.wav | grep Sampling -A1
Sampling rate                            : 44.1 kHz
Bit depth                                : 16 bits

Instead of fooling yourself with an oscilloscope, try to record a low level background noise (or sound) with a portable Olympus Hi-Res Audio Recorder. Then, you can amplify it with Audacity, and see (and hear) the difference.

But why do you need such a strong amplifier? Do you have problems with hearing?

igorzwx · Off-topic

Audacity FFT spectrogram has nothing to do with the quality of my sound card (Intel HDA codec).
The default minimum is -90dB. This is below the audible level.
_https://support.audacityteam.org/audio-analysis/spectral-analysis#plot-spectrum

After amplification, you can hear the song, but the level of rather unpleasant noise is very high.
But this because, perhaps, the song was recorded with the help of PulseAudio.

igorzwx · Off-topic

Yes, you may need an amplifier, especially for high frequencies, if you have problems with hearing.

igorzwx · Off-topic

If you open this wave with Audacity, you see nothing there, and Audacity's spectrogram shows nothing.
This may mean, perhaps, that human beings are not expected to hear anything.
However, if you amplify it with Audacity tool (50dB), you can hear the song and a lot of noise.
But it is still below -43dB. You can amplify it more, but then you may have a horrible noise as well.
It is, indeed, a sort of popular song. I do not know "who sings what".
I would not pass a test on popular songs.

igorzwx · Off-topic

Over 5% of the world’s population – or 430 million people – require rehabilitation to address their disabling hearing loss (including 34 million children). It is estimated that by 2050 over 700 million people – or 1 in every 10 people – will have disabling hearing loss.
_https://www.who.int/news-room/fact-sheets/detail/deafness-and-hearing-loss

Linux reached 4.03 percent of global market share in February, according to data from research firm Statcounter. That takes Linux past the 3 percent milestone it reached in June 2023. While we’re still far from the Year of the Linux Desktop, interest in Linux has somewhat grown lately.
_https://arstechnica.com/gadgets/2024/03/linux-continues-growing-market-share-reaches-4-of-desktops/

15 years ago, the data were different: deaf people - 2%, and, correspondingly, Linux Desktop users - 2%.

It seems to be a result of "natural selection" caused by PulseAudio. Therefore, it should not be difficult to verify these data with mathematical modeling.

The officially official Devuan Forum!

#226 Re: Freedom Hacks » A scientific test for resamplers. » 2024-07-13 22:06:22

#227 Re: Freedom Hacks » A scientific test for resamplers. » 2024-07-13 16:33:10

#228 Re: Freedom Hacks » A scientific test for resamplers. » 2024-07-13 15:06:52

#229 Re: Freedom Hacks » A scientific test for resamplers. » 2024-07-13 14:15:27

#230 Re: Freedom Hacks » A scientific test for resamplers. » 2024-07-13 08:05:17

#231 Freedom Hacks » A scientific test for resamplers. » 2024-07-11 21:03:53

#232 Re: Freedom Hacks » DXD waves and flacs (and dsf files) with OSS4 and ALSA » 2024-07-10 14:45:56

#233 Re: Freedom Hacks » DXD waves and flacs (and dsf files) with OSS4 and ALSA » 2024-07-10 11:18:54

#234 Re: Freedom Hacks » DXD waves and flacs (and dsf files) with OSS4 and ALSA » 2024-07-10 07:51:37

#235 Re: Freedom Hacks » DXD waves and flacs (and dsf files) with OSS4 and ALSA » 2024-07-09 09:48:30

#236 Freedom Hacks » DXD waves and flacs (and dsf files) with OSS4 and ALSA » 2024-07-08 21:34:34

#237 Re: Off-topic » How World Leaders are Scrambling to Secure Food in the Shadows » 2024-07-01 16:19:24

#238 Re: Off-topic » How World Leaders are Scrambling to Secure Food in the Shadows » 2024-07-01 15:48:25

#239 Re: Off-topic » How World Leaders are Scrambling to Secure Food in the Shadows » 2024-07-01 14:48:02

#240 Re: Off-topic » How World Leaders are Scrambling to Secure Food in the Shadows » 2024-07-01 14:11:16

#241 Re: Off-topic » Thoughts on Pipewire » 2024-06-30 20:32:55

#242 Re: Off-topic » Thoughts on Pipewire » 2024-06-30 19:50:54

#243 Re: Off-topic » Thoughts on Pipewire » 2024-06-30 14:12:04

#244 Re: Off-topic » Statistics is a very exact science » 2024-06-30 10:44:37

#245 Re: Off-topic » Statistics is a very exact science » 2024-06-30 09:33:48

#246 Re: Off-topic » Statistics is a very exact science » 2024-06-30 06:33:10

#247 Re: Off-topic » Statistics is a very exact science » 2024-06-29 21:04:35

#248 Re: Off-topic » Statistics is a very exact science » 2024-06-29 15:59:45

#249 Re: Off-topic » Statistics is a very exact science » 2024-06-29 15:41:46

#250 Off-topic » Statistics is a very exact science » 2024-06-29 13:37:24

Board footer