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1
3rd Party Plugins - (fb2k) / Re: Library Tree Discussion
Last post by yeyo -
@yeyo

If possible I would like to keep an auto check of features.

So can we have one more go before I'll likely make it a manual setting in panel properties (which only Wine users etc. whose systems don't suport ShowHtmlDialog will have to set on a one time basis)?

1) In v1.3.2 beta would you mind replacing popupbox.js in ...foo_spider_monkey_panel\packages\{E85C9EF0-778B-46DD-AF20-F4BE831360DD}\scripts with the @regor  version above to see if that fixes the issue?
v1.3.2 beta + replacing popupbox.js,Right click option still cannot be opened normally
2) If it still fails to open options, can you insert the following return true on line 53, after isHtmlDialogSupported() {,  of the @regor popupbox.js to bypass the check & confirm that the options then opens correctly.
Code: [Select]
	isHtmlDialogSupported() {
return true;
Use code:
Code: [Select]
	isHtmlDialogSupported() {
return true;
Replace the original code(52-54 line):
Code: [Select]
	isHtmlDialogSupported() {
if (this.soFeat.checked) return this.soFeat.gecko && this.soFeat.clipboard;
else this.soFeat.checked = true;
Can work normally

My operating system platform:
edition:Windows 11 professional 22H2
Operating system version:22621.160

3
WavPack / Re: Clicks/pops occur at the beginning of DSD wv tracks when playing as PCM
Last post by rutra80 -
I tend to think about PCM vs DSD, having two things on my mind:

1. PCM dynamic range is linear across whole frequency range, while in DSD it is not (low frequencies have far more dynamic range than ultrasonic ones which fall down to 1bit equivalent for 1411.2kHz@DSD64).

2. Both signals have bitrate, for stereo DSD64 it's what, 5644.8 kbps?

Converting DSD to PCM, kind of averages dynamic range.
So while keeping that 5644.8kbps, we could convert DSD64 to either:

- 16bit 176.4kHz PCM (that would keep quite a lot of loud noise at ultrasonic frequencies)

or

- 64bit 44,1kHz PCM (which maybe would be better - overshot dynamic range and almost no ultrasonics to deal with)


DSD would be interesting if it could stay like that in the whole audio chain - from recording to speakers. But it often gets converted to PCM sooner than later, making it quite useless IMO. Thats why I'd rather convert it to lower sample rate while keeping higher bit depth.
5
General - (fb2k) / [Linux] Avoiding re-sampling of Foobar playback?
Last post by Atronach -
Hello,

I'm struggling to bypass Wine re-sampling when playing music with Foobar on Linux.

I have set PulseAudio to avoid resampling unless necessary in /etc/pulse/daemon.conf:
Code: [Select]
avoid-resampling = true

Moreover, as my default audio device, I have set the "plug" plugin which also avoids unnecessary re-sampling in /etc/asound.conf:
Code: [Select]
pcm.!default { 
type plug
slave {
pcm "hw:PCH,0"
}
}
...you can see there the audio stream is sent straight to the subdevice 0 on the HDA Intel PCH device, which is 3.5 mm audio jack output on the PC in my case. This is my HW devices list:
Code: [Select]
david@david-opensuse:~> aplay -l
**** List of PLAYBACK Hardware Devices ****
card 0: PCH [HDA Intel PCH], device 0: Generic Analog [Generic Analog]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
card 0: PCH [HDA Intel PCH], device 3: HDMI 0 [HDMI 0]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
card 0: PCH [HDA Intel PCH], device 7: HDMI 1 [HDMI 1]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
card 0: PCH [HDA Intel PCH], device 8: HDMI 2 [HDMI 2]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
card 0: PCH [HDA Intel PCH], device 9: HDMI 3 [HDMI 3]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
card 0: PCH [HDA Intel PCH], device 10: HDMI 4 [HDMI 4]
  Subdevices: 1/1
  Subdevice #0: subdevice #0

My HW codec is capable of 44.1-192 kHz sampling rates and 16-24 bit depths:
Code: [Select]
david@david-opensuse:~> cat /proc/asound/card0/codec#0
Codec: Realtek Generic
Address: 0
AFG Function Id: 0x1 (unsol 1)
Vendor Id: 0x10ec0283
Subsystem Id: 0x80862068
Revision Id: 0x100003
No Modem Function Group found
Default PCM:
    rates [0x560]: 44100 48000 96000 192000
    bits [0xe]: 16 20 24
...
My audio chip is integrated Intel:
Code: [Select]
david@david-opensuse:~> lspci | grep -i audio
00:1f.3 Audio device: Intel Corporation Sunrise Point-LP HD Audio (rev 21)

So according to this I should get a bit-perfect output in most cases, or at least avoid the unnecessary re-sampling when the HW allows for it. And it works like this indeed with native Linux audio players like Cantata or Sayonara. If I play some hi-res audio track, I can see the stream is sent to the HW device with unchanged sample-rate:
Code: [Select]
david@david-opensuse:~> cat /proc/asound/card0/pcm0p/sub0/hw_params 
access: MMAP_INTERLEAVED
format: S32_LE
subformat: STD
channels: 2
rate: 96000 (96000/1)
period_size: 960
buffer_size: 19200
...also sample format consists of 32 bits so it's rather padded with zeroes than stripped of any bits of the source 24 bits samples so it's supposed to be OK.

I can set my Linux player apps to either Alsa or PulseAudio output and the result is the same.

Unfortunatelly, it doesn't work like this with Wine/Foobar. When I freshly install Foobar to a Wine prefix, winepulse.drv audio driver is used so Wine forwards the audio stream from Foobar to the PulseAudio output. This is what Foobar sees in its Playback -> Output -> Device list:

If I select one of the *[exclusive] variants, I get no sound. If I select either "Primary Sound Driver" or "Pulseaudio", I get some sound but the same 96 kHz source as before is down-sampled to 44.1 kHz:
Code: [Select]
david@david-opensuse:~> cat /proc/asound/card0/pcm0p/sub0/hw_params 
access: MMAP_INTERLEAVED
format: S32_LE
subformat: STD
channels: 2
rate: 44100 (44100/1)
period_size: 48000
buffer_size: 96000

I tried to switch Wine from using the winepulse.drv driver to the winealsa.drv one with winetrics: Select the default wineprefix -> Change settings -> sound=alsa. The playback output device list in Foobar changed to this:

Unfortunatelly, the result is similar - *[exclusive] variants provide no sound and "Primary Sound Driver", "Out: HDA Intel PCH - Generic Analog" and "Out: default" cause down-sampling to 48 kHz for a change:
Code: [Select]
david@david-opensuse:~> cat /proc/asound/card0/pcm0p/sub0/hw_params 
access: MMAP_INTERLEAVED
format: S32_LE
subformat: STD
channels: 2
rate: 48000 (48000/1)
period_size: 480
buffer_size: 1920

Is there someone who's been able to setup Foobar in Wine on Linux to bypass re-sampling of hi-res sources or to achieve bit-perfect output in general and so can advice me on how to do it?

Foobar version: 1.6.11
Wine version: 6.0-bp153.1.173
Linux distro: openSUSE Leap 15.3

Thanks.
7
3rd Party Plugins - (fb2k) / Re: Biography Discussion
Last post by kahel -
Since I've installed the version 1.3.3, I've noticed that my 'channel spectrum panel' was freezing every now and then...
Still happen with the 1.3.2... but doesn't with the 1.3.1, except the usual, 'normal' freeze while images first load.

The version 1.3.1 work fine, so no rush needed... just letting you know in case you could think of change that could affect performance... I've tested with 'filmstrip' and 'autocycle' disable... the freeze seem to happen every few second... without any image change... maybe something update in the background...

Might be worth exploring in the future... if I'm not the only one noticing those freeze...
To test... the easiest way would be in a music visualizer for example... also seem to impact performance while I scrolls down through my music list... not positive about that since it's a lot harder to notice in that case.
10
WavPack / Re: Clicks/pops occur at the beginning of DSD wv tracks when playing as PCM
Last post by bennetng -
To stay somewhat on-topic (foo_input_sacd and wavpack) yet still somewhat off-topic (about the audible click), here are some illustrations. The DSD test signal used was directly modulated from 2822400Hz, not upsampled from a lower sample rate. This signal is then decimated to 352.8kHz via different converters. When foo_input_sacd was used, all settings are using the 64-bit float version.

The -1dB@25kHz.txt filter I made can be download here:
https://archimago.blogspot.com/2022/06/notes-on-dac-dsd-1-bit-pdm-measurements.html?showComment=1655906640637#c3453242716296098960

These are the ones with more dramatic differences:
X
X
X


The last three screenshots are pretty close to the inherent noise floor of the modulator itself. It is kind of hard to visually differentiate some of the minor differences in some graphs even though one of them is using 24-bit output. Better download the pictures and use a photo viewer to rapidly switch back and forth to compare them. Of course, I've never claim audible differences, so don't ask for ABX log and such.
XXX