I use JRMC as well and see the same thing with DirectSound/no resampling. As a test I'm using a Hag USB/SPDIF converter on my XP box, Hag output is going to the SPDIF input on my RME card on a different machine. With nothing playing the RME software reports 48khz, as soon as I play a 44.1khz track it reports 44.1khz.
The Hag is limited to 48khz max. If I have time this weekend I can move the RME card to the XP box and just loop the SPDIF I/O and see what happens, although I'm reasonably certain of the outcome.
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Hi!
First I apologize for what can be a question has has been most probably answered/debated elsewhere, but at the moment I am finding many answers so I just get confused, so, I'd like to ask here, if it can be answered from your personal experience:
Which player would you recommend for a Linux user (Ubuntu or Linux Mint), that is equivalent (or as good and versatile as) Foobar?
I use an oldish PC with this soundcard: Creative Labs Audigy 2
Thanks and regards
Vix
First I apologize for what can be a question has has been most probably answered/debated elsewhere, but at the moment I am finding many answers so I just get confused, so, I'd like to ask here, if it can be answered from your personal experience:
Which player would you recommend for a Linux user (Ubuntu or Linux Mint), that is equivalent (or as good and versatile as) Foobar?
I use an oldish PC with this soundcard: Creative Labs Audigy 2
Thanks and regards
Vix
I do not have a reference for this, other than too many years using Windows. I tried everything, including turning Windows sounds off. The only way I could avoid K-mixer resampling was to use ASIO4All or the KS (Kernel Streaming) plugin for Foobar. I assume the Kernel Streaming plugin also allows the application's audio stream to be "primary" rather than taking a back seat to the windows sound system.
Indeed. Very possible. I have to admit that it has been some time since I have even bothered with all of this stuff because ultimately my solution was to abandon Windows altogether and use Linux, which allows much more control and far more options in quality players and DSPs.
Based on my understanding of the reference you cited earlier, I would expect the output to be 96kHz or 192kHz, respectively. However, based on my experience with the specific hardware I was using when I was using Windows XP as the source OS, I would expect to see 48kHz output.
On my current Linux system and with my current DAC, I would expect to see a rate of 44.1kHz, regardless of what the max rate of my DAC is. If I push a rate that my DAC cannot handle, I expect to see an error message, which is exactly the behaviour I want because it tells me that nothing in the system is making decisions on SRC. I want to determine precisely if/when SRC is done and what DSP is used for that SRC (Secret Rabbit Code, SoX, Shibatch SRC, etc.).
This is interesting (and contrary to my experience), but not interesting enough to get me to go back to Windows and try other hardware. As discussed above, I use Linux now and I have all the control I want/need.
Based on this and the posts by ra7, I suspect that my issue really must have been driver related. The thing is, when I was researching this some time ago, I found many, many people with the same Windows SRC issues (using various hardware) and the answer in the forums was always something to the effect of "Windows always resamples to 48kHz; use ASIO".
I will defer to those, like yourself, who somehow overcame this issue before going crazy (or perhaps you just got lucky).
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My two favourites right now for sheer configurability and sound quality are DeadBeef and Audacious, but if you want something that offers cataloguing like Foobar, go for Clementine.
By the way, are you using line-level (analog) out or SPDIF --> DAC ?
The reason I ask is because you sould be aware of the following spec on your sound card --> "SPDIF output up to 24-bit at 48 or 96kHz"
What this indicates is that, regardless of OS choice, any 44.1kHz files (probably the vast majority of what you listen to) will need to be resampled to 48kHz when you output via SPDIF. You may want to take control of this yourself by configuring your player (or plugin(s)) to do the SRC up to 48kHz. By default, ALSA will do this for you, but if you choose "direct hardware output with no software conversions" in your output plugin then you will get an error message instead (which leaves it up to you to choose whatever SRC plugin you want).
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A bit more info: One of the most flexible players with respect to tayloring "which sample rates are converted to what" is Audacious. Simply go to the "Output" menu and go to "Effects" and then check the "Sample Rate Converter" checkbox. Then go to "Output" --> "Effects" again and click the "settings..." entry under "Sample Rate Converter". This brings up an interface which allows you to define a specific, individual rate mapping for the following input rates: 8k, 16k, 22.05k, 44.1k, 48k, 96k, and 192k. It also allows you to select between the following interpolators: Linear, ZOH, Fastest Sync, Medium Sync, and Best Sync. Based on the specs of your sound card or DAC, and the power of your PC, you can make the best of your SRC. For example, if I was using your sound card's SPDIF interface on a lower powered machine, I would configure the following:
8k --> 48k
16k --> 48k
22.05k --> 48k
44.1k --> 48K
48k --> 48k
96k --> 96k
192 --> 96k
Medium Sync interpolator
If you have a soundcard, SPDIF interface, DAC, etc. that can do any rate up to 192, then this is a non-issue, of course.
Audacious also includes a "LADSPA Host" plugin in the "Output" --> "Effects" menu, which gives you access to dozens of excellent, open source LADSPA DSP plugins.
The plugins must be installed in the system separately and then they show up in the LADSPA configuration interface. To get a whole bunch without much effort, do:
sudo apt-get install tap-plugins swh-plugins invada-studio-plugins-ladspa caps
These four packages will give you more DSPs than you'll ever need. Some funky ones like "Vynil" and some "tube" simulators, etc.
8k --> 48k
16k --> 48k
22.05k --> 48k
44.1k --> 48K
48k --> 48k
96k --> 96k
192 --> 96k
Medium Sync interpolator
If you have a soundcard, SPDIF interface, DAC, etc. that can do any rate up to 192, then this is a non-issue, of course.
Audacious also includes a "LADSPA Host" plugin in the "Output" --> "Effects" menu, which gives you access to dozens of excellent, open source LADSPA DSP plugins.
The plugins must be installed in the system separately and then they show up in the LADSPA configuration interface. To get a whole bunch without much effort, do:
sudo apt-get install tap-plugins swh-plugins invada-studio-plugins-ladspa caps
These four packages will give you more DSPs than you'll ever need. Some funky ones like "Vynil" and some "tube" simulators, etc.
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One of the problems here is that the XP mixer is indeed screwed up. Either KS or ASIO is required to avoid SRC. The problem was fixed in Vista. But so many people avoided Vista for a number of issues real or perceived and never found out that exclusive mode and DS work correctly. And worse, these people extrapolate the XP problems into 7. Use exclusive mode and it doesn't matter what you use, DS, KS (which is very buggy when used with Vista/7) or ASIO.
As for Linux/Wine, I have no experience, but from what I read on the Foobar forum, Wine has some issues.
Bob
As for Linux/Wine, I have no experience, but from what I read on the Foobar forum, Wine has some issues.
Bob
The default EQ DSP in DeadBeef is Naoki Shibata's 18-band "SuperEQ". I find this to be an excellent EQ.
As Shibata-san explains:
"An accurate equalizer for winamp 2(yes, it's old). It uses FIR filter with 16383 taps, which is substantially longer than other equalizers. Although it has many knobs to change frequency characteristics, it does not degrade audio quality."
I have to agree. The audio quality seems to be completely unaffected by this EQ. In fact, this is one of the main reasons why I currently prefer DeadBeef over Audacious. Audicious' SRC plugin is extremely flexible (as discussed in my earlier post) but DeadBeef has the advantage of using Shibata-san's fantastic SSRC (Shibatch Sample Rate Converter). It can only be configured to apply a single output rate to all input rates, but it is an extremely high quality SRC and it is very fast. I use this when I want to play anything higher than 48kHz (my USB DAC only does up to 48, but I am going to be addressing that issue shortly).
If you don't like DeadBeef for some reason and/or prefer Audacious for other reasons (SRC configuration flexibility), you can use one of the LADSPA DSP EQs (lots of them provided in the plugin packages that I mentioned). I didn't look at all of them, but one of them has 15 bands for sure and works quite well. There are also various parametric EQ DSPs, bandpass filter DSPs, crossover filter DSPs, etc. etc. I did not try very many of them because I just really like DeadBeef (and the fact that the 18-band SuperEQ plugin is nicely integrated into the DeadBeef interface).
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Foobar-in-WINE is unnecessary and suboptimal. There's no doubt about that.
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Yes, probably. I've seen soundcard drivers that forced everything to 48K. Fortunately my present one does not. It's not perfect, tho. Put into 96Khz mode it will force any lower sample rate to 48Khz. I don't know were the resampling takes place, tho.
For me XP and Direct Sound work great. Bit perfect if I want it.
This would drive me nuts... But that's just me.
Off Topic:
A pirate walks into a bar with a steering wheel in his pants.
Bartender: "What's with the steering wheel in your pants?"
Pirate: "Arrrrr, it's driving me nuts!"
In computer architecture perspective, the highest standard requirement of Digital Signal Processing (DSP) for audiophile is recap as below:
1) “Bit Perfect” audio condition
2) Stereo 24Bit/192KHz Studio Master quality
3) Speed of audio streaming at least 1.12 MB/s (i.e. 2x24x192,000=9,216,000 bps)
Since the operating system of PC does not design for audiophile purpose, the general guildlines to meet the captioned three conditions are listed as below:
Condition#1: It requires by-pass for both hardware and software. Hardware bypass is the SRC (Sample Rate Converter) in audio codec by software driver. Software bypass is the original audio mixer by software driver to direct route from Application software to hardware.
Condition#2: It depends on the three factors of source, interface and DAC chipset to support up to 24Bit/192KHz Studio Master quality
Condition#3: It depends on the speeds of CPU and computer bus
Take Microsoft Window O/S as the platform for example, the following two configurations are proved to true to fulfill the captioned three conditions.
1) CPU: Intel Prentium III or above for both desktop and notebook computers
2) O/S: Window XP or above
3) Audio codec: HD Audio codec (built-in on board)
4) Media player: Footbar2000 with ASIO4All plug-in (To bypass Window KMixer)
1) CPU: Intel Prentium III or above for old desktop computer
2) O/S: Window XP or above
3) Audio codec: AC97 Audio codec (built-in on board)
4) Media player: Footbar2000 with ASIO4All plug-in (To bypass Window KMixer)
5) Add-on Interface: Creative Labs Audigy 2 PCI sound card (To replace AC97 bypass SRC)
For detail explanation, please refer to my [ARTICLE] here in which I have just finished.
1) “Bit Perfect” audio condition
2) Stereo 24Bit/192KHz Studio Master quality
3) Speed of audio streaming at least 1.12 MB/s (i.e. 2x24x192,000=9,216,000 bps)
Since the operating system of PC does not design for audiophile purpose, the general guildlines to meet the captioned three conditions are listed as below:
Condition#1: It requires by-pass for both hardware and software. Hardware bypass is the SRC (Sample Rate Converter) in audio codec by software driver. Software bypass is the original audio mixer by software driver to direct route from Application software to hardware.
Condition#2: It depends on the three factors of source, interface and DAC chipset to support up to 24Bit/192KHz Studio Master quality
Condition#3: It depends on the speeds of CPU and computer bus
Take Microsoft Window O/S as the platform for example, the following two configurations are proved to true to fulfill the captioned three conditions.
1) CPU: Intel Prentium III or above for both desktop and notebook computers
2) O/S: Window XP or above
3) Audio codec: HD Audio codec (built-in on board)
4) Media player: Footbar2000 with ASIO4All plug-in (To bypass Window KMixer)
1) CPU: Intel Prentium III or above for old desktop computer
2) O/S: Window XP or above
3) Audio codec: AC97 Audio codec (built-in on board)
4) Media player: Footbar2000 with ASIO4All plug-in (To bypass Window KMixer)
5) Add-on Interface: Creative Labs Audigy 2 PCI sound card (To replace AC97 bypass SRC)
For detail explanation, please refer to my [ARTICLE] here in which I have just finished.
I want to add a remark that some people may classify the sound of music with American style, British style, Asian style, etc… This DSP approach is not classified for any style. It just moves the source code exactly from Master Recording Studio at 24Bit/192KHz to the home for broadcasting without any third party treatment. If the source code is recorded with American singer, then music is American style as well as recorded with British singer, then British style and so on. In contrast, it simulates the situation that this DSP approach moves you from home to sit at Master Recording Studio for listening to the singer with real live effect.
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