Dev1 Galaxy Forum / Resampling and the Gibbs phenomenon with Audacity

Re: Resampling and the Gibbs phenomenon with Audacity

dummy@example.com (igorzwx) — Mon, 22 Jul 2024 09:09:20 +0000

aluma wrote:

I asked you to post information that I had repeatedly posted myself for my system.

I hope that I am mistaken, but it looks like you are simulating "problems with ALSA", in order to convince people to use pulseaudio.

Re: Resampling and the Gibbs phenomenon with Audacity

dummy@example.com (aluma) — Mon, 22 Jul 2024 07:25:06 +0000

If you do not believe my words, why should I believe yours?

I asked you to post information that I had repeatedly posted myself for my system.
But I don’t care at all, I’ll leave it at that.

Re: Resampling and the Gibbs phenomenon with Audacity

dummy@example.com (igorzwx) — Sun, 21 Jul 2024 19:34:02 +0000

Firefox is playing video in youtube:

$ firefox 2>&1
[Child 3342, MediaDecoderStateMachine #1] WARNING: 7fa31a74d4c0 Could not set cubeb stream name.: file ./dom/media/AudioStream.cpp:321
Input:  44100 Hz, 2 ch, 's32_le' (0xa): dummy = 0, period = 1764
Output: 48000 Hz, 2 ch, 's16_le' (0x2): dummy = 0, period = 1920
Rates:  44100 --> 48000 (J: 0.00%, T: FFT, W: Vorbis)
Ok.

FFT is the fftrate resampler. It is GPL3, the source code is here:
_https://github.com/PetrovSE/fftrate

$ fuser -av $(find /dev/snd -type c 2>/dev/null)
                     USER        PID ACCESS COMMAND
/dev/snd/controlC0:  igor       2229 F.... mate-settings-d
                     igor       2324 F.... mate-volume-con
/dev/snd/pcmC0D0c:
/dev/snd/pcmC0D0p:   igor       3175 F...m firefox-esr
/dev/snd/seq:
/dev/snd/timer:      igor       3175 f.... firefox-esr

It works without apulse, and it works with apulse as well

$ apulse firefox 2>&1
Input:  48000 Hz, 2 ch, 's32_le' (0xa): dummy = 0, period = 1920
Output: 48000 Hz, 2 ch, 's16_le' (0x2): dummy = 0, period = 1920
Rates:  48000 --> 48000 (J: 0.00%, T: None, W: Planar)
Ok.

Notice that the resampler is not active (T: None).
It seems that it makes sense to run browsers with apulse.
It helps to improve sound quality with Min, Brave, Chromium and Chrome.
It might be a bug in ALSA backend.

The same video with Chromium:

$ apulse chromium 2>&1
Input:  44100 Hz, 2 ch, 's32_le' (0xa): dummy = 0, period = 1764
Output: 48000 Hz, 2 ch, 's16_le' (0x2): dummy = 0, period = 1920
Rates:  44100 --> 48000 (J: 0.00%, T: FFT, W: Vorbis)
Ok.

$ fuser -av $(find /dev/snd -type c 2>/dev/null)
                     USER        PID ACCESS COMMAND
/dev/snd/controlC0:  igor       2229 F.... mate-settings-d
                     igor       2324 F.... mate-volume-con
/dev/snd/pcmC0D0c:
/dev/snd/pcmC0D0p:   igor       7968 F...m firefox-esr
                     igor       8428 F...m chromium
/dev/snd/seq:
/dev/snd/timer:      igor       7968 f.... firefox-esr
                     igor       8428 f.... chromium

If you do not believe my words, why should I believe yours?
It might be very difficult to believe that you really want to configure ALSA.

Re: Resampling and the Gibbs phenomenon with Audacity

dummy@example.com (aluma) — Sun, 21 Jul 2024 17:37:11 +0000

I also do not have sound devices defined in the Firefox settings (about:support).
And, nevertheless, sound works with "pure alsa".

These are again just words. Screenshot, as in my thread at the link above, details (output of the corresponding bash commands) of the operating system in the studio.

Re: Resampling and the Gibbs phenomenon with Audacity

dummy@example.com (igorzwx) — Sun, 21 Jul 2024 17:01:03 +0000

aluma wrote:

With pure alsa, the sound was everywhere, except for the browser, which “did not see” the sound system. Either pipewire or jack was required.

If it works with pulseaudio, it does not mean that it cannot work with "pure alsa".

But I can agree that it might be difficult to create a correct ~/.asoundrc
That is why, I am using arateconf.

aluma wrote:

There is no sound device defined in the FF Daedalus settings.

I also do not have sound devices defined in the Firefox settings (about:support).
And, nevertheless, sound works with "pure alsa".

Re: Resampling and the Gibbs phenomenon with Audacity

dummy@example.com (aluma) — Sun, 21 Jul 2024 16:45:40 +0000

My post #4

What our posts have in common is that they have nothing to do with this forum itself, the forum of the Devuan distribution, and your links and advice on proprietary formats and paid resamples in no way correspond to Freedom Hacks, as I understand it.

Re: Resampling and the Gibbs phenomenon with Audacity

dummy@example.com (chris2be8) — Sun, 21 Jul 2024 16:16:36 +0000

Could some moderator please move this thread to Off-topic ? It has nothing to do with Devuan.

Re: Resampling and the Gibbs phenomenon with Audacity

dummy@example.com (aluma) — Sun, 21 Jul 2024 16:12:26 +0000

If you start a new topic, and I will try to help.

Thank you very much, no help needed.

https://dev1galaxy.org/viewtopic.php?id=6118

Re: Resampling and the Gibbs phenomenon with Audacity

dummy@example.com (igorzwx) — Sun, 21 Jul 2024 12:50:04 +0000

On Devuan Daedalus 5.0, sound works in Firefox out of the box, after the removal of pulseaudio
_https://dev1galaxy.org/viewtopic.php?pid=49837#p49837

The problem might be that you want to use your Audigy2, and it is not the default device.
The simplest solution to such problems is to run

arateconf

and select Audigy2 as "default device".
The manual is here:

ALSA without PulseAudio and PipeWire
_https://dev1galaxy.org/viewtopic.php?id=6644

If you start a new topic, and I will try to help.

NOTE: It is not necessary to remove PulseAudio. You can remove it later, when ALSA is configured and sound works with

apulse firefox

Some useful commands for testing ALSA, you may find here:
_https://dev1galaxy.org/viewtopic.php?pid=50114#p50114
_https://wiki.archlinux.org/title/Advanced_Linux_Sound_Architecture

Re: Resampling and the Gibbs phenomenon with Audacity

dummy@example.com (aluma) — Sun, 21 Jul 2024 11:54:58 +0000

What does bad manners have to do with it?
Look through the forum, I asked for advice about sound in Firefox. With pure alsa, the sound was everywhere, except for the browser, which “did not see” the sound system. Either pipewire or jack was required.
I have 5 different computers at home with the same Linux and DE and the sound settings are different for everyone.

Re: Resampling and the Gibbs phenomenon with Audacity

dummy@example.com (igorzwx) — Sun, 21 Jul 2024 11:34:21 +0000

Of course, it is very bad manners to remove pulseaudio or pipewire.
The good manners is to use pulseaudio, pipewire, or any other crap, because of loyalty to authorities.

Re: Resampling and the Gibbs phenomenon with Audacity

dummy@example.com (aluma) — Sun, 21 Jul 2024 11:20:29 +0000

Just words without any objective data, measurements, etc.
And about “pulseaudio and/or pipewire” - this is nonsense. Their necessity is determined by specific conditions, sound card, BIOS, etc.

Re: Resampling and the Gibbs phenomenon with Audacity

dummy@example.com (igorzwx) — Sun, 21 Jul 2024 10:02:46 +0000

Yes, I am using my method, and it works.
If the goal is to improve sound quality on your computer, you can simply remove pulseaudio and/or pipewire, install Petrov's fftrate plugin, and configure it for 32bit 192kHz (Intel HDA), or other HiRes format, if it is supported by your sound card. For USB headsets, you may try 16bit 48kHz.

Re: Resampling and the Gibbs phenomenon with Audacity

dummy@example.com (aluma) — Sun, 21 Jul 2024 08:14:32 +0000

Have you tried it yourself?
You won't improve anything and you won't hear any difference from resampling your favorite Petrov.
Empty chatter.

Re: Resampling and the Gibbs phenomenon with Audacity

dummy@example.com (igorzwx) — Sun, 21 Jul 2024 08:03:38 +0000

If the goal is to improve sound quality on your notebook, my method may work.

What is important is to prevent resampling and format conversion.
Otherwise, you may not know, what you are actually testing.