A scientific test for resamplers.

igorzwx · 2024-07-11 21:03:53

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

Last edited by igorzwx (2024-07-11 21:50:26)

aluma · 2024-07-13 04:19:57

Maybe these numbers mean something to someone, but not to me.

So, the scientific approach (like a pro in measurement systems):

The goal - is an objective comparison of the effect on the sound of different resamplers and different systems (ALSA, pipeware, pulseaudio, etc.).
Method - instrumental measurements.

This requires another computer with a more or less high-quality sound card (I use audigy2, 24 bit, 195 kHz), an input/output connecting wire and two programs:
- RightMark Audio Analyzer (rmaa6) for measuring all parameters of the sound card being tested.
- SpectraLab or Linux analog for measuring distortion of specific signals.

Further, there are many possibilities, for example, by recording a pure sinusodal signal from a generator with a frequency of 1 kHz and measuring the distortion when it is played back in different versions of the system, we can objectively evaluate the comparison of alsa vs pipeware vs pulseaudio. Or various resamplers.

The photo shows my primitive cord and an example of wasted work - the electrostatic screen for the Yamaha 724 sound card, according to the measurement results, turned out to be absolutely useless.

igorzwx · 2024-07-13 08:05:17

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?

aluma · 2024-07-13 11:46:30

Can you honestly say why you are raising these questions?
Except out of a desire to be known as an original?

We have already found out that, like the vast majority of ordinary users, you hear 14 of the standard 16 bits. Why are you 32?

Real Hi-Fi and Hi-End systems require very significant costs and often a separate, acoustically treated room. Few people can afford it, and even more people don’t need it at all. Because the goal of all this, like any art, is to convey emotions, and for this, some people don’t even need sound, notes are enough.

It's pointless to try to "hear" the Hi-Fi and Hi-End level without the appropriate equipment, it's like trying to see the landscape through a dirty window.

steve_v · 2024-07-13 12:59:36

Further to^

If you cannot hear the difference between Petrov's dct and fft resamplers

Then any further testing is but a curiosity, because at the end of the day, differences you can't hear are irrelevant for anything you intend to listen to.

Human hearing sucks, and the vast majority of final reproduction equipment (i.e. speakers and their environment) sucks as well... because so long as the latter suck significantly less than the former, nobody will hear it and it doesn't matter.
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.

igorzwx · 2024-07-13 14:15:27

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?

pl · 2024-07-13 14:26:17

igorzwx wrote:

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

Ahem, but ain't that true? What's the point if *no* difference is heard? To unnecessarily occupy more disk space or what?

aluma · 2024-07-13 14:41:34

@igorzwx
Yes, to be sure of the freshness of the fish, you need to try it.
I talk about my practical conclusions during the next audition. About the multi-bit DAC, as opposed to the 32-bit delta-sigma in the mentioned article.
You're talking about other people's opinions.
If you don’t want to understand, it in detail, with measurements, check the sound quality of different programs, it’s up to you. Of course it’s easier to say that the rest are deaf.

Last edited by aluma (2024-07-13 14:42:59)

igorzwx · 2024-07-13 15:06:52

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?

chris2be8 · 2024-07-13 16:18:13

The real test is if you can tell which system is which in a blind test. If you have two systems, one cheap and one top quality, set up with a switch to control which one feeds the speakers, can you tell which switch setting is which system?

To be really thorough try it several times, with a friend changing which one is which. And a few control runs where both switch settings get the same system.

igorzwx · 2024-07-13 16:33:10

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.

Last edited by igorzwx (2024-07-13 17:53:56)

steve_v · 2024-07-13 21:08:12

it was detected by an OSS4/Gentoo user in a "blind test"

(emphasis mine) One anecdotal sample is barely a "test" at all, and not even remotely scientific.
Then you go on to claim that blind tests have problems, because the usual "bit-perfect" "exclusive mode" audiophool nonsense.

As for "old theory", I've been there, done that, (as well as built and tested plenty of proper HiFi gear) so I'm no longer gullible enough to get sucked into such arguments... I'm out of here, have fun.

igorzwx · 2024-07-13 22:06:22

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 · 2024-07-14 13:17:14

aluma wrote:

I use audigy2, 24 bit, 195 kHz

Perhaps, it is Audigy2, 24 bit, 192 kHz.

aluma wrote:

There are many possibilities, for example, by recording a pure sinusodal signal from a generator with a frequency of 1 kHz and measuring the distortion when it is played back in different versions of the system, we can objectively evaluate the comparison of alsa vs pipeware vs pulseaudio. Or various resamplers.

It seems that your experiments can be simplified and made more exact.
All sorts of sinusoidal signals can be easily created with Audacity (with mathematical exactitude).

sudo apt install audacity

Audacity → Generate → Tone → select frequency and amplitude

You can also change resolution:

Audacity → Edit → Preferences → Quality → "Default Sample Rate" and "Default Sample Format"

Then you may try to measure the distortions in playback. It might be very interesting.

But you may better start a new topic for this.

aluma · 2024-07-15 05:47:11

"Before you make a discovery, look at the textbook." (C) (Not mine)
Have an understanding of the theory of harmonic steam processes and the capabilities of SpectraLab.
And Audigy2, yes, 192 kHz (4*48).
Thank you, it’s my fault, there are a lot of buttons, I’m the only one, who couldn’t cope.

P.S. I didn't expect it myself.

Yesterday, upon requesting “SpectraLab...” Google returned a link to one of the radio forums with my participation from 2011.
My point is that these issues were discussed a long time ago and everything has already been clarified.

And you can imagine my “interest” in empty chatter on this topic...

Last edited by aluma (2024-07-15 07:21:12)

igorzwx · 2024-07-15 09:31:02

If everything has already been clarified, then, perhaps, you may answer my question.
May I ask it 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?

CONNECTING THE MPD-8 TO A WINDOWS BASED PC or APPLE MAC
Supported resolutions: PCM: all sample rates up to 384kHz and word lengths up to 24 bits.
DSD: native up to 11.2MHz and DoP up to 5.6MHz
Connection protocol and method: USB 2.0 / Audio class 2.0 compliant, asynchronous mode where MPD-
8 generates high precision clock master and computer is clock slaved.
PC requirements: Windows 7 or later, ASIO compatible player software such as JRiver
etc.
Mac requirements: OS X (10.6.6 or later)
If your computer is a Mac running OSX 10.6.6 or later you don’t need any driver software as OSX already supports all PCM formats up to 384kHz sample rate.
_http://playbackdesigns.com/docs/manuals/PlaybackDesigns_MPD-8_Manual1v1.pdf

$ mediainfo "08-Faust - Funeral March Of A Marionette - 32bit.wav" | grep Audio -A11
Audio
Format                                   : PCM
Format profile                           : Float
Codec ID                                 : 3
Codec ID/Hint                            : IEEE 
Duration                                 : 4 min 35 s
Bit rate mode                            : Constant
Bit rate                                 : 22.6 Mb/s
Channel(s)                               : 2 channels
Sampling rate                            : 352.8 kHz
Bit depth                                : 32 bits
Stream size                              : 743 MiB (100%)

I've written here and there about my preference for listening to 32-bit PCM versus 24-bit PCM files (either 352.8kHz or 384kHz)...
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/

If he is using MAC, DXD waves and flacs are likely to be downsampled to 48kHz (default) . OSX resamplers can do the work.

If "OSX already supports all PCM formats up to 384kHz sample rate", it does not mean that the OSX driver for MPD-8 DAC supports 384kHz sample rate. It may simply mean that OSX resamplers support 384kHz sample rate.
In this case, Foobar2000 may fail to play DXDs in "exclusive mode".

_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.

The result of downsampling depends on the quality of the resampler and resolution of the original file.
If the original DXD wave is 32bit Float, downsampling to 48kHz (or 96kHz) may produce a better sound quality.
That is why, perhaps, Rushton Paul claims that 32bit DXDs are better than 24bit DXDs.

Would you agree with this?

Last edited by igorzwx (2024-07-15 16:16:48)

aluma · 2024-07-15 16:48:38

I don't want to speculate about something I haven't heard.

In general, the higher the sampling frequency, the “smaller” the steps, by smoothing which the DAC receives an analog signal. All the more subtle nuances we can hear. In practice, there is, for example, jitter, unevenness of the count caused by a change in the frequency of the clock generator. The latter is possible, for example, due to impulse noise in the power supply, interference, etc. and so on.
How does it differ in different operating modes of the resampler and different processor loads?

So I just don't know.

P.S. By analogy with the landscape outside the window. The sound reproduction system consists of different parts and the quality of each determines the degree of “cleanliness of the window”. The super quality of one element can simply be hidden by the mediocre quality of another. There are many options in each specific case.

Last edited by aluma (2024-07-15 17:11:47)

igorzwx · 2024-07-15 17:45:13

OK, let us put it as simple as possible.

Can your DAC play 32bit float audio files (which it should not play)?

Have you tried?

Last edited by igorzwx (2024-07-15 17:54:07)

aluma · 2024-07-15 17:50:38

I have no idea, I don't have such files

igorzwx · 2024-07-15 17:55:39

You can download free samples here:
_https://positive-feedback.com/reviews/music-reviews/what-we-hear-with-dxd-32-bit-files/

You can play them with MPlayer

sudo apt install mplayer

$ mplayer "08-Faust - Funeral March Of A Marionette - 32bit.wav"
MPlayer UNKNOWN-12 (C) 2000-2023 MPlayer Team
do_connect: could not connect to socket
connect: No such file or directory
Failed to open LIRC support. You will not be able to use your remote control.

Playing 08-Faust - Funeral March Of A Marionette - 32bit.wav.
libavformat version 59.27.100 (external)
libavformat file format detected.
[wav @ 0x7fab12c87d00]Discarding ID3 tags because more suitable tags were found.
[lavf] stream 0: audio (pcm_f32le), -aid 0
[lavf] stream 1: video (png), -vid 0, Picture
VIDEO:  [MPNG]  607x603  0bpp  90000.000 fps    0.0 kbps ( 0.0 kbyte/s)
libva info: VA-API version 1.17.0
libva info: Trying to open /usr/lib/x86_64-linux-gnu/dri/iHD_drv_video.so
libva info: Found init function __vaDriverInit_1_17
libva error: /usr/lib/x86_64-linux-gnu/dri/iHD_drv_video.so init failed
libva info: va_openDriver() returns 1
libva info: Trying to open /usr/lib/x86_64-linux-gnu/dri/i965_drv_video.so
libva info: Found init function __vaDriverInit_1_8
libva info: va_openDriver() returns 0
==========================================================================
Opening video decoder: [ffmpeg] FFmpeg's libavcodec codec family
libavcodec version 59.37.100 (external)
[png @ 0x7fab121fdd20]Requested frame threading with a custom get_buffer2() implementation which is not marked as thread safe. This is not supported anymore, make your callback thread-safe.
Selected video codec: [ffpng] vfm: ffmpeg (FFmpeg PNG)
==========================================================================
Clip info:
 artist: Alexander Gibson -RPHO
 date: 2024-01-15
 album: Gounod - Bizet - Alexander Gibson Royal Opera House Orchestra
 encoder: Merging Technologies Album Publishing
 title: Faust - Funeral March Of A Marionette
 track: 8/9
Load subtitles in ./
==========================================================================
Opening audio decoder: [pcm] Uncompressed PCM audio decoder
AUDIO: 352800 Hz, 2 ch, floatle, 22579.2 kbit/100.00% (ratio: 2822400->2822400)
Selected audio codec: [pcm] afm: pcm (Uncompressed PCM)
==========================================================================
AO: [oss] 192000Hz 2ch s16le (2 bytes per sample)
Starting playback...
Could not find matching colorspace - retrying with -vf scale...
Opening video filter: [scale]
Movie-Aspect is 1.01:1 - prescaling to correct movie aspect.
[swscaler @ 0x7fab12e13f20]bicubic scaler, from rgba to yuv420p using MMXEXT
VO: [vdpau] 608x604 => 609x604 Planar YV12 
Movie-Aspect is 1.01:1 - prescaling to correct movie aspect.
VO: [vdpau] 608x604 => 609x604 Planar YV12 
No pts value from demuxer to use for frame!
pts after filters MISSING
A:   0.0 V:-9223372036854775808.0 A-V:  0.000 ct:  0.000   0/  0 ??% ??% ??,?% 0 0 0.54x 
Possibly bad interleaving detected.
Use -ni option if this causes playback issues and avoid or fix the program that created the file.
A:   3.0 V:-9223372036854775808.0 A-V:  0.000 ct:19449.693   0/  0  0%  0%  0.0% 0 0 0.54x 

Exiting... (Quit)

It does play DXDs (with downsampling and format conversion):

==========================================================================
Opening audio decoder: [pcm] Uncompressed PCM audio decoder
AUDIO: 352800 Hz, 2 ch, floatle, 22579.2 kbit/100.00% (ratio: 2822400->2822400)
Selected audio codec: [pcm] afm: pcm (Uncompressed PCM)
==========================================================================
AO: [oss] 192000Hz 2ch s16le (2 bytes per sample)
Starting playback...

Last edited by igorzwx (2024-07-15 18:09:43)

aluma · 2024-07-15 17:58:29

You're all talking about these, of course not.

igorzwx · 2024-07-15 18:13:14

I can play them with MPlayer (Devuan) on Intel HDA (see my previous post).

Last edited by igorzwx (2024-07-15 18:14:29)

aluma · 2024-07-15 18:16:13

Yeah, “that Caruso is nonsense, my neighbor sang it to me yesterday...”

igorzwx · 2024-07-15 18:56:43

Do you think that "Caruso" may sound better with MPD-8 DAC, when it is downsampled to 48kHz?

The assumption is that Rushton Paul is using Mac for playing 32bit Float DXDs with the Playback Designs MPD-8 DAC (which cannot play 32bit Float DXDs).

Audio Device Setup for Mac
_https://www.sweetwater.com/sweetcare/articles/audio-midi-setup-for-mac/

MacBook Pro (2023)
Applications > Utilities > Audio MIDI Setup

MacBook Pro Microphone (built-in)
1 channel
Bit rate: 32bit Float
Default sample rate: 48kHz
Maximum sample rate: 96kHz

MacBook Pro Speakers (built-in)
2 channels
Bit rate: 32bit Float
Default sample rate: 48kHz
Maximum sample rate: 96kHz

32-bit float audio can capture the absolutely ludicrous range of up to 1,528 dB. That’s not only massively beyond the scope of 24-bit audio, but it’s beyond the scale of what even counts as a sound on Earth.
_https://www.wired.com/story/32-bit-float-audio-explained/

It means, perhaps, that might be possible to record "The Year 1812, Solemn Overture, Op. 49"

The 1812 Overture is scored for an orchestra that consists of the following:
Brass band: "Open" instrumentation consisting of "any extra brass instruments" available. In some indoor performances, the part may be played on an organ. Military or marching bands also play this part. Note: the brass band or its substitute is meant to play during the finale only.
Woodwinds: 1 piccolo, 2 flutes, 2 oboes, 1 cor anglais, 2 clarinets in B♭ and 2 bassoons
Brass: 4 horns in F, 2 cornets in B♭, 2 trumpets in E♭, 3 trombones (2 tenor, 1 bass) and 1 tuba
Percussion: timpani, orchestral bass drum, snare drum, cymbals, tambourine, triangle, carillon
Strings: violins I & II, violas, cellos and double basses.
Artillery: one battery of cannon, or even ceremonial field artillery.
_https://en.wikipedia.org/wiki/1812_Overture#Instrumentation

Last edited by igorzwx (2024-07-16 16:37:31)

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#1 2024-07-11 21:03:53

A scientific test for resamplers.

#2 2024-07-13 04:19:57

Re: A scientific test for resamplers.

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Re: A scientific test for resamplers.

#4 2024-07-13 11:46:30

Re: A scientific test for resamplers.

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Re: A scientific test for resamplers.

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Re: A scientific test for resamplers.

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Re: A scientific test for resamplers.

#8 2024-07-13 14:41:34

Re: A scientific test for resamplers.

#9 2024-07-13 15:06:52

Re: A scientific test for resamplers.

#10 2024-07-13 16:18:13

Re: A scientific test for resamplers.

#11 2024-07-13 16:33:10

Re: A scientific test for resamplers.

#12 2024-07-13 21:08:12

Re: A scientific test for resamplers.

#13 2024-07-13 22:06:22

Re: A scientific test for resamplers.

#14 2024-07-14 13:17:14

Re: A scientific test for resamplers.

#15 2024-07-15 05:47:11

Re: A scientific test for resamplers.

#16 2024-07-15 09:31:02

Re: A scientific test for resamplers.

#17 2024-07-15 16:48:38

Re: A scientific test for resamplers.

#18 2024-07-15 17:45:13

Re: A scientific test for resamplers.

#19 2024-07-15 17:50:38

Re: A scientific test for resamplers.

#20 2024-07-15 17:55:39

Re: A scientific test for resamplers.

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Re: A scientific test for resamplers.

#22 2024-07-15 18:13:14

Re: A scientific test for resamplers.

#23 2024-07-15 18:16:13

Re: A scientific test for resamplers.

#24 2024-07-15 18:56:43

Re: A scientific test for resamplers.

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