Hi folks.
I'd like to share a new little project with you guys.
Since I am more sitting at my desk then I am sitting in front of my stereo,
I thought why don't you fix yourself a nice "nearfield setup".
I selected some active pro-audio monitors ( Emes Black, K&H O100, Adam P11A)
for testing, since all of them have a real good reputation and run in the - at least to me, affordable 1300$ league. ( Yeah - I know - this time no DIY speakers!)
I just got the Emes delivered, I am waiting for the other ones to arrive.
I fired them up with some nice power-cords and mains-filters and connected
them to the headphone-output for the time being. I am running an internal
Soundmax HD from Analog Devices on my T60p Thinkpad.
The setup was quite easy. I just had to triple the buffer sizes compared to
my usb-dac.
The nice thing about the monitors is that you can adjust them on the input gain.
You're able to run the players almost at full speed (100% volume), which avoids
losses on the digital side.
I did a first soundcheck on the different sw-players using the Emes Black:
Ranking:
1. brutus
2. xmms-realtime
3. XX-Highend under XP engine 1
4. JRiver with ASIO4ALL
It again proves to me that Linux is in the lead. And the new thing here - not only on USB devices!!
The Emes ( Coaxes , ideal for nearfield purposes) are doing really well. The effort is extremly low to get it going and it's really fun to listen to. The Emes are so neutral,
without loosing resolution or dynamics, you can listen to them for hours.
I just played a nice piece from Marta Gomez. The sound is delivered extremely clean and nicely presented stagewise. It almost makes you feel as if you were face to face with her. Real nice experience.
This you'll neither experience on a normal stereo nor on headphones.
BUT
One issue I still have to resolve, perhaps somebody can help.
The linux hda_intel driver seems to generate quite some noise ( already at boot-up). Much more then I experience under Windows. My alsa-mixer PCM and master pots are at 100%. I am actually using direct device access hw:0,0 to pass dmix by.
Somehow the Alsa volume controls (master/pcm) are still working.
I am wondering if the soundcard comes with an own mixer, which is handled by the alsa mixer instead of going through dmix.
Does somebody have an idea of a. how to get rid of the noise and b. how to avoid any mixer in the loop? Or any other ideas to get the issues solved?
One more: With the second delivery I also get a Benchmark DAC1 for testing - let see how this piece performs. I'll let you know.
Cheers
\Klaus
I'd like to share a new little project with you guys.
Since I am more sitting at my desk then I am sitting in front of my stereo,
I thought why don't you fix yourself a nice "nearfield setup".
I selected some active pro-audio monitors ( Emes Black, K&H O100, Adam P11A)
for testing, since all of them have a real good reputation and run in the - at least to me, affordable 1300$ league. ( Yeah - I know - this time no DIY speakers!)
I just got the Emes delivered, I am waiting for the other ones to arrive.
I fired them up with some nice power-cords and mains-filters and connected
them to the headphone-output for the time being. I am running an internal
Soundmax HD from Analog Devices on my T60p Thinkpad.
The setup was quite easy. I just had to triple the buffer sizes compared to
my usb-dac.
The nice thing about the monitors is that you can adjust them on the input gain.
You're able to run the players almost at full speed (100% volume), which avoids
losses on the digital side.
I did a first soundcheck on the different sw-players using the Emes Black:
Ranking:
1. brutus
2. xmms-realtime
3. XX-Highend under XP engine 1
4. JRiver with ASIO4ALL
It again proves to me that Linux is in the lead. And the new thing here - not only on USB devices!!
The Emes ( Coaxes , ideal for nearfield purposes) are doing really well. The effort is extremly low to get it going and it's really fun to listen to. The Emes are so neutral,
without loosing resolution or dynamics, you can listen to them for hours.
I just played a nice piece from Marta Gomez. The sound is delivered extremely clean and nicely presented stagewise. It almost makes you feel as if you were face to face with her. Real nice experience.
This you'll neither experience on a normal stereo nor on headphones.BUT
One issue I still have to resolve, perhaps somebody can help.
The linux hda_intel driver seems to generate quite some noise ( already at boot-up). Much more then I experience under Windows. My alsa-mixer PCM and master pots are at 100%. I am actually using direct device access hw:0,0 to pass dmix by.
Somehow the Alsa volume controls (master/pcm) are still working.
I am wondering if the soundcard comes with an own mixer, which is handled by the alsa mixer instead of going through dmix.
Does somebody have an idea of a. how to get rid of the noise and b. how to avoid any mixer in the loop? Or any other ideas to get the issues solved?
One more: With the second delivery I also get a Benchmark DAC1 for testing - let see how this piece performs. I'll let you know.
Cheers
\Klaus
This is the most intressting thread in DIYAUDIO for me in the last two years! I have learned a lot! Thank you all who is contributing to a very nice thread about hi-fidelity sound in Linux.
I am a fan of Linux and I use it alot, but am not that good at it yet. Tying to learn.
I just want to show you my resent setup that works quite good.
Its Ubuntu 7.10 rc2 (released today) and Foobar2000 + dsp-crossover running under WINE. I hope to be able to tweak this in the same way you have, with a realtime kernel and so forth.
Probably not the most good sounding, but I like the easy way of playing with the crosssover. I hope I can make something realy good out of it.
Cheers Johannes.
I am a fan of Linux and I use it alot, but am not that good at it yet. Tying to learn.
I just want to show you my resent setup that works quite good.
Its Ubuntu 7.10 rc2 (released today) and Foobar2000 + dsp-crossover running under WINE. I hope to be able to tweak this in the same way you have, with a realtime kernel and so forth.
Probably not the most good sounding, but I like the easy way of playing with the crosssover. I hope I can make something realy good out of it.
Cheers Johannes.
Hi folks.
I think I found an answer to the noise issue.
I looked up the specs of the soundmax card.
It seems - as I guessed - that I am running into the soundcard mixer with the
linux controls and not into dmix. That's good,
The high noisefloor on the 100% setting of the alsamixer-PCM slider might be explained because I am adding 12db gain to the input signal of the soundcard
if the alsamixer-slider stands on 100%.
The soundcard mixer range goes from +12 to -34,5db gain resp. attentuation
according to the spec.
My conclusion: alsamixer-pcm setting 100% means +12db gain to the noisefloor.
Unfortunately these poor alsamixer slider doesn't have a scale. I put it
down approx. 25% to get down to 0db. Here we go.
Side-note: The specs of the card are looking really poor:
SNR is just 88db on the line out. Which is probably much better than the headphone out. THD line-out vs. headphone out is -92 vs. -75db.
BTW: The footnote on SNR says: "Guaranteed by design - Not production tested."
I think that's it. If somebody else has a better idea let me know.
Cheers
Klaus
I think I found an answer to the noise issue.
I looked up the specs of the soundmax card.
It seems - as I guessed - that I am running into the soundcard mixer with the
linux controls and not into dmix. That's good,
The high noisefloor on the 100% setting of the alsamixer-PCM slider might be explained because I am adding 12db gain to the input signal of the soundcard
if the alsamixer-slider stands on 100%.
The soundcard mixer range goes from +12 to -34,5db gain resp. attentuation
according to the spec.
My conclusion: alsamixer-pcm setting 100% means +12db gain to the noisefloor.
Unfortunately these poor alsamixer slider doesn't have a scale. I put it
down approx. 25% to get down to 0db. Here we go.
Side-note: The specs of the card are looking really poor:
SNR is just 88db on the line out. Which is probably much better than the headphone out. THD line-out vs. headphone out is -92 vs. -75db.
BTW: The footnote on SNR says: "Guaranteed by design - Not production tested."
I think that's it. If somebody else has a better idea let me know.
Cheers
Klaus
Hi folks.
I don't want to bother you again about these ALSA-mixer settings,
however I finally managed to set it exactly to 0db - Master and PCM.
It's actually not too difficult.
In the end my earlier guess regarding the added gain, if the slider stands on 100%
was confirmed.
You need to start the alsamixer from a shell. Just type
$ alsamixer
under "item" you'll see the exact gain values in db for the active slider.
Just put them all on 0db. Your soundcard should run pretty neutral this
way.
Make sure that you've got the mike turned off.
Cheers
\Klaus
I don't want to bother you again about these ALSA-mixer settings,
however I finally managed to set it exactly to 0db - Master and PCM.
It's actually not too difficult.
In the end my earlier guess regarding the added gain, if the slider stands on 100%
was confirmed.
You need to start the alsamixer from a shell. Just type
$ alsamixer
under "item" you'll see the exact gain values in db for the active slider.
Just put them all on 0db. Your soundcard should run pretty neutral this
way.
Make sure that you've got the mike turned off.
Cheers
\Klaus
alsa
Gents and Mams
Save your alsamixer settings by 'alsactl store n', where n is the soundcard number starting from 0. Obs! Different settings for root and normal user...
New version of Alsa 1.0.15 just released for those eager to test out. Will be included in next kernel release (2.6.23.2 or 2.6.24?).
New soundcards supported, optimizations and bug fixes.
Cheers,
Tom
Gents and Mams
Save your alsamixer settings by 'alsactl store n', where n is the soundcard number starting from 0. Obs! Different settings for root and normal user...
New version of Alsa 1.0.15 just released for those eager to test out. Will be included in next kernel release (2.6.23.2 or 2.6.24?).
New soundcards supported, optimizations and bug fixes.
Cheers,
Tom
Hi folks.
Just tried to hook-up the Benchmark Dac1 USB.
Unfortunately it doesn't seem to be supported. (The Linux Trap!)
I should have looked it up earlier. (Linux beginner mistake!)
The TAS1020 used in the DAC 1 is a very standard USB interface chip. Even MS supports it via standard usbaudio.sys in 24/96. I actually didn't really expected that the DAC is not supported via Linux.
Anybody with a TAS1020 DAC around who got it running under Linux?
Cheers
Just tried to hook-up the Benchmark Dac1 USB.
Unfortunately it doesn't seem to be supported. (The Linux Trap!)
I should have looked it up earlier. (Linux beginner mistake!)
The TAS1020 used in the DAC 1 is a very standard USB interface chip. Even MS supports it via standard usbaudio.sys in 24/96. I actually didn't really expected that the DAC is not supported via Linux.
Anybody with a TAS1020 DAC around who got it running under Linux?
Cheers
Hi folks.
The Benchmark DAC1 is running. It sounds great with the ADAM P11As.
Much better then any MS setup so far.
Somehow I managed to get it to work with aplay. I still have problems with
XMMS and brutefir.
Today I also tried 2.6.23 with the kamikaze3 patch-set. The patch-set includes
some ported patches of Con Kolivas. Unfortunately it is not using the
staircase scheduler of Con - it comes with CFS of Ingo Molnar.
In the end I listened to the same muddy CFS sound I listened before.
I should have done something else and could have saved some hours.
Our 2.6.22 CK setup sounds ways better.
BTW - Just to mention it again.
The Benchmark is clearly sensitive to the incoming signal jitter! It sounds great if the PC setup is perfect.
Now I can prepare for the DDDAC/Benchmark comparision.
Cheers
Klaus
The Benchmark DAC1 is running.
It sounds great with the ADAM P11As.Much better then any MS setup so far.
Somehow I managed to get it to work with aplay. I still have problems with
XMMS and brutefir.
Today I also tried 2.6.23 with the kamikaze3 patch-set. The patch-set includes
some ported patches of Con Kolivas. Unfortunately it is not using the
staircase scheduler of Con - it comes with CFS of Ingo Molnar.
In the end I listened to the same muddy CFS sound I listened before.
I should have done something else and could have saved some hours.
Our 2.6.22 CK setup sounds ways better.
BTW - Just to mention it again.
The Benchmark is clearly sensitive to the incoming signal jitter! It sounds great if the PC setup is perfect.
Now I can prepare for the DDDAC/Benchmark comparision.
Cheers
Klaus
Interesting to hear your DAC comparison Soundcheck...
I have not tried the Kamikaze3 patch, what are the main benefits?
I have tried 2.6.23.1 @10.000HZ by manually patching kernel files my self, and my first impression is that it is on par with ck, that's promising... 2.6.23.1 @ 1.000 is not on par, so Hz do matters a big deal soundwize... If you want to check it out, see which files the ck patch changes on 2.6.22, and do it manually your self.
Cheers,
Tom
I have not tried the Kamikaze3 patch, what are the main benefits?
I have tried 2.6.23.1 @10.000HZ by manually patching kernel files my self, and my first impression is that it is on par with ck, that's promising... 2.6.23.1 @ 1.000 is not on par, so Hz do matters a big deal soundwize... If you want to check it out, see which files the ck patch changes on 2.6.22, and do it manually your self.
Cheers,
Tom
TBM said:
Hi Tom.
Here is the link. There you'll see what's inside the kamikaze3.
http://forums.gentoo.org/viewtopic-t-600890-postdays-0-postorder-asc-start-0.html
It's as easy to apply as ck. I do have some acpi issues (fan-control
doesn't work) and do not see any of the "tweaking-variables" I used to tweak.
Perhaps I missed out something ( for sure the 10000Hz setting
). If you're saying that it sounds as good as ck afterwards I need to have a closer look at it. I'll put my .config file in the wiki.
Perhaps than it's easier to discuss certain issues like dyntics,
irqbalancing asf. later on.
Cheers
Klaus
SRC
Hi folks.
Kind of off-topic:
I'd like to refer to a nice sample rate conversion comparision:
http://src.infinitewave.ca/
As you know I am upsampling my material to 48khz.
As you'll see the Secret Rabbit Code in "Best Sinc" mode, the one I am using, is performing quite well considering that it is a freeware converter.
Even the big ones such as Steinberg are not any better.
One thing which caught my attention though is the iZotope 64-bit
SRC. It seems to be slightly better than SRC.
I am wondering if it is available for testing purposes.
I am just checking it out.
Cheers
\Klaus
Hi folks.
Kind of off-topic:
I'd like to refer to a nice sample rate conversion comparision:
http://src.infinitewave.ca/
As you know I am upsampling my material to 48khz.
As you'll see the Secret Rabbit Code in "Best Sinc" mode, the one I am using, is performing quite well considering that it is a freeware converter.
Even the big ones such as Steinberg are not any better.
One thing which caught my attention though is the iZotope 64-bit
SRC. It seems to be slightly better than SRC.
I am wondering if it is available for testing purposes.
I am just checking it out.
Cheers
\Klaus
sangram said:
Best of luck -- I've installed the previous release and found it relatively smooth going. Had a bit of trouble with monitor resolution at first, so make sure you select your monitor's maximum resolution at the start of installation. Saves troubleshooting it later on.
Another tip is if the buttons on software like Audacity do not appear properly, it could be because you chose 'india' as your language option/locale during the install. Took me a while to figure that one out but you can change this option after install and mine works fine now.
I also found this install guide for v7.04 to be helpful. The same site now has a guide for standard ubuntu v.7.10 which also applies to Ubuntu studio.
Hope it works out -- I'm really impressed with Ubuntu Studio 7.04, both in terms of sound quality and general usability. When I get the chance I'll change to v7.10 for the realtime kernel.
ssmith said:
It didn't. My soundcard was not detected correctly or supported, and the ethernet controller isn't working either, so I couldn't get any updates for the soundcard either. I had a beautiful desktop but nothing to do with it... I have a bit of experience with Gnome and X, but none with installing drivers from tarballs.
Back to XP, will continue trying to troubleshoot the Ubuntu install over the next few days. I'm kind of hopeless at the software, so I did the install on a backup disk anyway. Nothing lost if it doesn't work out.
I believe a lot of people are having issues with internet connections in 7.10, hope to see a slightly more noob-friendly solution than some lines of text, or a binary for the 1.0.15 ALSA which should get my soundcard to work. I don't really need an internet connection except for the stupid update process. Why can't I just download and transfer and install in synaptic?
SunRa said:
Hi.
1000Hz is the internal clock frequency for setting the interrupt IRQ clock.
It means - the process issuing an IRQ gets a timeslot of 1ms to run its task. (This is mainly the limiting factor on MS machines,
when it comes to latencies below 1ms. )
As you know I was looking for reduction of latencies in the system.
I thought 1ms is probably not needed for most of the interrupts on a pure audio machine.
I changed it from 1kHz to 5 and ended up at 10khz.
( This option was delivered by the CK patch only)
The system runs quite stable at 10khz. The 10khz clock is generating more overhead, still the audio process
runs a lot smoother.
Tom and Eddie are confirming it.
There is one problem with it.
Somehow I can hear the low volume 10kHz tone when sitting very close to the PC.Cheers
\Klaus
sangram said:
Hi.
I just looked it up.
http://www.alsa-project.org/main/index.php/Matrix:Vendor-Creative_Labs
1212m is supposed to run with ALSA 1.0.14
1212m PCI V2 is supposed to run with ALSA 1.0.15
your driver is snd-emu10k1
try from a terminal:
$ sudo modinfo snd-emu10k1
and
$ sudo lsmod | grep snd-emu10k1
and
$ cat /proc/asound/cards
It'll show if your card is recognised.
ALSA 1.0.15 has been released 5 days ago. You'll see it earliest beginning of next year in an official Ubuntu package I'd guess.
In the end the best would be to get used of howto generate a kernel by yourself. This way you are usually a couple of month ahead of Ubuntu. Installing a new kernel incl. ALSA is IMO easier
then trying to install ALSA manually.
Anyhow. You need to google your problems. Usually you'll find an answer to your problems very quick.
Cheers
Klaus
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