Re: [Alsa-user] intel hda vs asus xonar STX
Hi Pierre Lorenzon!
On 2015.09.27 at 19:13:17 +0200, Pierre Lorenzon wrote next:
> In fact I expected such an answer but I would like not use jack
> or pulse if not necessary. If the asus xonar essence stx is
> able to mix sources there's no need to use more only enable it.
Xonar STX ("AV100" chip which is basically the same as C-media 8787)
does not support mixing multiple sources. So you have to either use
pulseaudio or rely on something like dmix.
(jack works too but I wouldn't recommend it unless you need low latency
audio in professional audio-related applications, it can be a headache
when used for typical multimedia tasks with random crappy software that
wants to play audio)
Note that if you have first model of *STX*, not ST or second STX model
(actually I'm not sure about second STX), for quality reasons you'd want
to avoid 44100/88200/176400 audio sample rates, so instead of default
pulse configuration when it tries to pick audio rate to avoid conversion
(e.g. you're playing single stream of 44100 music in your audio player -
it sets card audio rate to 44100 because card supports it), you should
lock the rate to 96000 or 192000 instead, e.g.
default-sample-format = s24le
default-sample-rate = 96000
in pulseaudio's daemon.conf
Lower audio quality and lower SNR in 44100 and multipies of it is quite
audible. However, this only applies to PCI-E model (STX), PCI model (ST)
has no problems with any rates.
--
Vladimir
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Re: [Alsa-user] Alsa simultaneous use of usb-sound card - BW problem
Hi Ralf Mardorf! On 2014.08.04 at 19:05:37 +0200, Ralf Mardorf wrote next: On Mon, 2014-08-04 at 19:53 +0300, Burak METİN wrote: When i use these cards with different usb buses, it is ok but i need to use them in same bus. IIRC the plug'n'play audio USB driver is limited to USB1, for USB2 proprietary drivers are needed. But perhaps I'm confusing something. Assumed I shouldn't be mistaken, then perhaps the bandwidth really is to small. Can't be true.. Here is example of USB 2.0 audio device (bcdUSB: 2.00), completely plug and play: Bus 002 Device 005: ID 20b1:0002 XMOS Ltd Device Descriptor: bLength18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 239 Miscellaneous Device bDeviceSubClass 2 ? bDeviceProtocol 1 Interface Association bMaxPacketSize064 idVendor 0x20b1 XMOS Ltd idProduct 0x0002 bcdDevice3.30 iManufacturer 1 XMOS iProduct2 XMOS USB Audio 2.0 iSerial 3 bNumConfigurations 2 -- Vladimir -- Infragistics Professional Build stunning WinForms apps today! Reboot your WinForms applications with our WinForms controls. Build a bridge from your legacy apps to the future. http://pubads.g.doubleclick.net/gampad/clk?id=153845071iu=/4140/ostg.clktrk ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] teac UD-H01, bad 24 bit playback
Hi Rutger Noot! On 2013.12.20 at 08:23:55 +0100, Rutger Noot wrote next: I didn't make any progress on this, except for finding out on the web that the TEAC has some firmware quirks (supposed to be handled by recent kernels). It appears that the teac driver is indispensable in windows to get the device running usb 2.0. Is there a way to find out (under linux) if the audio device is configured as usb 2.0 ? The usb subsystem also detects a HID device (which is usb 1.0 ??). If the audio is treated as usb 1, this would explain a lot. Run lsusb -v, find your device in output - bcdUSB parameter is version. E.g. Bus 002 Device 005: ID 20b1:0002 XMOS Ltd Device Descriptor: bLength18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 239 Miscellaneous Device bDeviceSubClass 2 ? bDeviceProtocol 1 Interface Association bMaxPacketSize064 idVendor 0x20b1 XMOS Ltd idProduct 0x0002 bcdDevice3.30 iManufacturer 1 XMOS iProduct2 XMOS USB Audio 2.0 ... (Ignore iProduct here - that's just device name, but as bcdUSB shows it's working in 2.0 mode) By the way, USB 1.1 has quite enough bandwith for 24 bit / 96 kHz output. I did that on my older M-Audio Audiophile USB. It's just that you can't go higher than that, or can't record + playback at the same time. USB 2.0 is only strictly required if you need 24/192 or more than 2 channels for 24/96 or more than 4 channels for 24/48. -- Vladimir -- Rapidly troubleshoot problems before they affect your business. Most IT organizations don't have a clear picture of how application performance affects their revenue. With AppDynamics, you get 100% visibility into your Java,.NET, PHP application. Start your 15-day FREE TRIAL of AppDynamics Pro! http://pubads.g.doubleclick.net/gampad/clk?id=84349831iu=/4140/ostg.clktrk ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
[Alsa-user] Creative X-Fi Titanium and 24/96k on digital output
Hello everybody. I'm trying to get Creative X-Fi Titanium PCIe to output 96000 audio through digital output (optical) and fail to do it. System in question is Fedora 17 with kernel-3.4.6-2.fc17.x86_64 kernel and ALSA 1.0.25. Driver is snd-ctxfi 1.03, device is 05:00.0 Audio device: Creative Labs X-Fi Titanium series [EMU20k2] (rev 04) dmesg|grep -i alsa gives [4.172004] ALSA sound/pci/ctxfi/ctatc.c:1295 ctxfi: chip 20K2 model SB0880 (1102:0042) is found [4.490065] ALSA sound/pci/ctxfi/cttimer.c:424 ctxfi: Use xfi-native timer Anyhow problem is that even though X-Fi is supposed to work in HD modes, I can only can get it to work at 16/48k mode. It has few devices $ grep name /proc/asound/card0/pcm*p/info /proc/asound/card0/pcm0p/info:name: Front/WaveIn /proc/asound/card0/pcm0p/info:subname: subdevice #0 /proc/asound/card0/pcm1p/info:name: Surround /proc/asound/card0/pcm1p/info:subname: subdevice #0 /proc/asound/card0/pcm2p/info:name: Center/LFE /proc/asound/card0/pcm2p/info:subname: subdevice #0 /proc/asound/card0/pcm3p/info:name: Side /proc/asound/card0/pcm3p/info:subname: subdevice #0 /proc/asound/card0/pcm4p/info:name: IEC958 Non-audio /proc/asound/card0/pcm4p/info:subname: subdevice #0 Trying to playback through any device except for 0,4 produces silence on digital output (is that correct behaviour? On SBLive and Audigy analog devices were routed to digital out, but it doesn't seem to be the case here; am I missing something?). But 0,4 device seem to work only in two modes: 16/44.1 and 16/48. If trying to open it in 16/96 mode or 24/48, 24/96, 32/48, 32/96 modes it doesn't work - I get invalid argument and other similar errors from speaker-test or other applications. When using plughw device or outputting through pulse and such output locks to 16/48 mode, as seen in /proc/asound/card0/pcm4p/sub0/hw_params Some googling suggested that reference_rate module parameter controls it, but it can be set only to 48k and 44k, not 96k. I'm using default values, reference_rate=48000 and multiple=2. I tried changing rate and other values with iecset but it produces no effect. -- Vladimir -- Live Security Virtual Conference Exclusive live event will cover all the ways today's security and threat landscape has changed and how IT managers can respond. Discussions will include endpoint security, mobile security and the latest in malware threats. http://www.accelacomm.com/jaw/sfrnl04242012/114/50122263/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] are cheap USB cards with analog line input supported by ALSA ?
Hi Sergei Steshenko! On 2012.04.30 at 02:03:34 +0400, Sergei Steshenko wrote next: You apparently lack imagination. Here is a number of hints for you: 1) how can you know how many input channels I need ? 3) how can you know whether my computer (which is not necessarily a desktop) at has Line In input ? Making lots of USB soundcards to work properly might be an issue by itself, even if each card works separately. There can be various issues with USB bus causing latency problems resulting in cracking and other problems if you actually try to use lots of USB soundcards. Reliable solutions for getting lots of line input on USB are using devices which are made for that; of course it won't be as cheap as a bunch of USB soundcards, but they are made for this job. There are various devices with these functions from M-Audio, E-MU and some other manufacturers. An example would be M-Audio Fast Track Ultra, USB 2.0 device which provides 8 analog inputs - I haven't checked if it works with ALSA, though, but I'm pretty sure it's possible to find something that works. That said, it's probably cheaper to buy lots of cheap USB single-channel soundcards from ebay or DX or something and check if they just work - and if they don't, think about real multichannel USB interfaces. But don't assume that they will all work together just because single one did - check if they actually work correctly when you feed required amount of input streams into them. -- Vladimir -- Live Security Virtual Conference Exclusive live event will cover all the ways today's security and threat landscape has changed and how IT managers can respond. Discussions will include endpoint security, mobile security and the latest in malware threats. http://www.accelacomm.com/jaw/sfrnl04242012/114/50122263/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Can't get Soundblaster X-Fi Titanium to work. What else should I try?
Hi d...@dougster.com! On 2012.03.02 at 15:13:27 -0700, d...@dougster.com wrote next: I'm trying to set up a RedHat 5, update 4, 64 bit workstation for audio playback. I found the X-Fi Titanium at a local store and checked the ALSA soundcard matrix which said that it is partially supported. I don't really need anything other than playback, so that seemed alright to me. I downloaded ALSA 1.0.25 and followed the INSTALL document, but when I get to step 8 and give the command modprobe snd-ctxfi I get the error: snd_ctxfi: Unknown symbol try_to_del_timer_sync. Redhat isn't exactly the best distro regarding multimedia support, and this is the cause of your problem. This card most likely isn't supported.. by downloading fresh alsa you might make it work but there are no promises - as you saw, it didn't help you. RHEL 5 update 4 is very old, and I doubt there is easy way of making it work, at least while sticking with that version. You might replace kernel, but there is no point in using Redhat then, as you will lose support by doing this. You might try to update to 5.8, but I'm really not sure if it's going to solve this problem. If you have support contract, try reinstalling system to RHEL 6.2. Redhat server licenses allow you to use any version - I haven't checked if it's the same with workstation licenses, but most probably it is. If you don't have active Red Hat support contract, then I might suggest you some free alternative like Scientific Linux 6.2 or CentOS 6.2 - if you prefer RHEL-compatible distro for one reason or another. They have more or less fresh X-Fi driver. If RHEL compatibility doesn't matter to you, you can pick any popular modern Linux distro, be that Fedora, Ubuntu, OpenSUSE, Mint or tons of others. As long as it's not too old, X-Fi will work just fine. I also tried downloading and building XFiDrv_Linux_Public_US_1.00.tar.gz from Creative, but when I run make, I get the error: cthw20k2.c:1315: error: implicit declaration of function 'mdelay'. That driver is even less supported, so don't follow this path at all :) -- Vladimir -- Virtualization Cloud Management Using Capacity Planning Cloud computing makes use of virtualization - but cloud computing also focuses on allowing computing to be delivered as a service. http://www.accelacomm.com/jaw/sfnl/114/51521223/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] plughw versus hw
Hi Pierre Habraken! On 2011.06.21 at 15:42:14 +0200, Pierre Habraken wrote next: Thanks to you (and to Vladimir) I understand better now why I need plughw to play 24bits files. If it's not taking too much of your time, I'd have one more question: When I try to play a 24bits/192KHz file using plughw, aplay -v displays the data below, without any error message, but nothing is sent to the DAC. Is this because the Xonar DX Linux driver does not take in account 192KHz sounds ? But in that case why Alsa does not report any error message ? Check your DAC specification, it probably can accept 192 khz streams only at 16 bit, and for 24 bits are limited to 96khz. It's quite typical and probably is S/PDIF bandwidth limitation (not sure); you have to use HDMI to get more. There is no error because 24/192 streams in itself work fine, and DAC chip on audio card can process that stream - it's just that there is trouble either sending it through S/PDIF or receiving by external DAC, which is outside of the scope of your card, so there is no error. -- Vladimir -- EditLive Enterprise is the world's most technically advanced content authoring tool. Experience the power of Track Changes, Inline Image Editing and ensure content is compliant with Accessibility Checking. http://p.sf.net/sfu/ephox-dev2dev ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] plughw versus hw
Hi Pierre Habraken! On 2011.06.21 at 16:07:51 +0200, Pierre Habraken wrote next: The contents of /proc/asound/card0/pcm1p/sub0/hw_params looks very similar to the one displayed by aplay -v. I assume that aplay gets its data from /proc... Yes, there is no need to look there if you are using aplay -v; it's handy when you are using something else to play and want to check card state. I don't know what exactly do pulseaudio and I don't like using things the operation of which I do not understand. Also, I read a lot of negative things about pulseaudio. So I've deactivated it. Thanks again for your help. Well, it's better to try for yourself instead of blindly turning it off :) Some people do have reasons to turn it off, for example because of buggy sound driver which causes troubles with pulse (which tries to use some advanced features), or because it interferes with some audio processing applications. But mostly it's just something that takes the blame, because it's default component of modern linux desktops and people often blame it even if the real cause of trouble is elsewhere. The main reason to use pulseaudio is comfort for desktop usage; mixing multiple software streams from various application with per-application volume controls that restores state over reboot and guaranteed audio quality regardless of streams that are played back (like, you can lock it to use your audio card in 24/96 mode only), and ability to move streams between outputs and cards (like analog/digital output, usb headset, bluetooth audio, sending audio over network etc) on the fly. You can check list of features in wikipedia http://en.wikipedia.org/wiki/PulseAudio#Features to see if there is something you like, or on official wiki page http://www.pulseaudio.org/wiki/AboutPulseAudio There is no reason to use pulse if you are satisfied with current setup, however lot of people like its features. -- Vladimir -- EditLive Enterprise is the world's most technically advanced content authoring tool. Experience the power of Track Changes, Inline Image Editing and ensure content is compliant with Accessibility Checking. http://p.sf.net/sfu/ephox-dev2dev ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] plughw versus hw
Hi Pierre Habraken! On 2011.06.20 at 19:32:28 +0200, Pierre Habraken wrote next: I can imagine that this is a FAQ, but I could not find a clear answer : which precise difference(s) distinguish(es) plughw and hw from each other ? Does plughw apply sound processing that hw does not ? plughw *might* apply simple sound processing if needed, mostly channels conversion and rate conversion if required. It doesn't have to apply processing. hw doesn't support such processing only works when operating strictly in mode that audio card support. If you have device that supports only 2 channel, 16 bit 48000 mode then hw device won't be able to playback 2/16/44100 stream, or mono stream for example; you'll get an error when you try. But plughw will accept such streams and do the conversion. However, if you use plughw and output 2/16/48000 stream then no conversion is needed and most likely plughw won't be doing any processing. Note that using both hw and plughw can lead to specific problems, so it's best to use default device unless you have very specific requirements. -- Vladimir -- EditLive Enterprise is the world's most technically advanced content authoring tool. Experience the power of Track Changes, Inline Image Editing and ensure content is compliant with Accessibility Checking. http://p.sf.net/sfu/ephox-dev2dev ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Hi, both nspluginwrapper and operapluginwrapper from Opera are not redirected to Pulseaudio as my configs state that they should
Hi Linux User! On 2010.01.21 at 11:12:48 -0500, Linux User wrote next: I'm on x86_64 system using a 32Bit Flash player and both plugin wrappers seem to bypass my configurations and go straight to my PCM device --- Do you have 32-bit pulse libs and 32-bit alsa plugin installed (32-bit libasound_module_pcm_pulse.so, libasound_module_ctl_pulse.so present in system, have all required libraries and work)? -- Vladimir -- Throughout its 18-year history, RSA Conference consistently attracts the world's best and brightest in the field, creating opportunities for Conference attendees to learn about information security's most important issues through interactions with peers, luminaries and emerging and established companies. http://p.sf.net/sfu/rsaconf-dev2dev ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Hi, both nspluginwrapper and operapluginwrapper from Opera are not redirected to Pulseaudio as my configs state that they should
Hi Linux User! On 2010.01.21 at 12:47:12 -0500, Linux User wrote next: Thanks for your quick response... I just ran a search and both were in there respective directories. I would have to say that I have all my required libraries and work fine as applications do connect to Pulseaudio, if they don't natively support Pulse, sooo. It's just those two and I don't know, why heck why... hmmm Only difference here should be the fact that you're using 32-bit plugins, that's why I asked if you really have 32-bit pulse system working. You should have 2 copies in your system, like -rwxr-xr-x 1 root root 24032 Мар 26 2008 /usr/lib64/alsa-lib/libasound_module_pcm_pulse.so -rwxr-xr-x 1 root root 20656 Мар 26 2008 /usr/lib/alsa-lib/libasound_module_pcm_pulse.so lrwxrwxrwx 1 root root17 Янв 25 2008 /usr/lib64/libpulse.so.0 - libpulse.so.0.4.0 lrwxrwxrwx 1 root root17 Янв 25 2008 /usr/lib/libpulse.so.0 - libpulse.so.0.4.0 Another caveat might be the way 64-bit opera executes 32-bit plugins code. Maybe it does something special which prevents pulse plugin from working. Try checking if 32-bit plugin works in 32-bit firefox, for example. I'm going to add a ~/.assoundrc and see if that remedy the issue =) On Thu, 21 Jan 2010 11:31:56 -0500, Vladimir Mosgalin mosga...@vm10124.spb.edu wrote: Hi Linux User! On 2010.01.21 at 11:12:48 -0500, Linux User wrote next: I'm on x86_64 system using a 32Bit Flash player and both plugin wrappers seem to bypass my configurations and go straight to my PCM device --- Do you have 32-bit pulse libs and 32-bit alsa plugin installed (32-bit libasound_module_pcm_pulse.so, libasound_module_ctl_pulse.so present in system, have all required libraries and work)? -- Using Opera's revolutionary e-mail client: http://www.opera.com/mail/ -- Throughout its 18-year history, RSA Conference consistently attracts the world's best and brightest in the field, creating opportunities for Conference attendees to learn about information security's most important issues through interactions with peers, luminaries and emerging and established companies. http://p.sf.net/sfu/rsaconf-dev2dev ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user -- Vladimir -- Throughout its 18-year history, RSA Conference consistently attracts the world's best and brightest in the field, creating opportunities for Conference attendees to learn about information security's most important issues through interactions with peers, luminaries and emerging and established companies. http://p.sf.net/sfu/rsaconf-dev2dev ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] M-Audio Fast Track support?
Hi Ng Oon-Ee! On 2009.11.24 at 06:31:00 +0800, Ng Oon-Ee wrote next: To find out the features etc, go to the Maudio web site. Not sure what details you want. I do know the features of the device, I was wondering how feature-complete the drivers in alsa were =) I have M-Audio Audiophile USB and all of its features are supported, including 4 x 4, AC3/DTS passthrough and 24/96 mode. I'm using analog digital outs all the time and digital in at times, haven't checked analog in but it should work too. I have no idea about MIDI support, it's said to work. 4 x 4 scheme is working and you can address any input output independently if you wish in that mode. You can read more details here http://www.mjmwired.net/kernel/Documentation/sound/alsa/Audiophile-Usb.txt All the features described there are in basic snd-usb-audio driver and I think that they should work with any similar device (just audiophile usb was most available usb audio interface from m-audio some time ago). USB cards usually don't require firmwares, unlike Firewire cards, though I might be wrong here, better check alsa site for details. -- Vladimir -- Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day trial. Simplify your report design, integration and deployment - and focus on what you do best, core application coding. Discover what's new with Crystal Reports now. http://p.sf.net/sfu/bobj-july ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] 96 kHz, 24 bit usb external soundcard
Hi HamRadio! On 2009.06.28 at 11:29:58 +0200, HamRadio wrote next: I'm looking for a 96 kHz - 24 bit external usb 2.0 soundcard well supported by linux/alsa drivers. Does exist such a piece of hardware or am I just dreaming? Thank you in advance for answering. Is usb 2.0 part important? M-Audio Audiophile USB is pretty nice external 96/24 usb 1.1 soundcard, well-supported (well I mean it's 2.0 probably but working at 11 mb/s). Catch is, 96/24 mode is half duplex. Only playback or only recording. 24/48 mode is full duplex and 16/48 mode provides 4 channels in, 4 channels out. You can read details in Audiophile-Usb.txt file from kernel documentation. -- Vladimir -- ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] 24-bit and 96KHz file
Hi Amaury De Ganseman! On 2008.09.03 at 13:16:42 +0200, Amaury De Ganseman wrote next: I have a 24-bit/96KHz file. When I play it, it says that it plays in s16_le and samplerate ouput is 48KHz (I see that on my hardware equaliser). I have an ESI Juli@ and use the alsa driver of the 2.6.26. ffmpeg doesn't have a (proper) support of 24-bit audio. Even when it can be read, it's trimmed to 16-bit for processing output. There seems to be some progress in this area; if 24-bit audio is important to you, ask for details in ffmpeg-users mailing list. -- Vladimir - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] asoundrc: dsnoop and capture mix
Hi Pete! On 2008.02.11 at 18:53:55 +0100, Pete wrote next: thanks for your answer. You're right: jack is simpler to configure. But I would like to make a configuration for users, wich is transparent so they just can use any alsa-application. So if anybody know's if there is an solution at all? You should try pulseaudio. alsa-pulse plugin works pretty well (at least much better than alsa-jack plugin), a lot of higher-level sound systems, for example SDL and gstreamer can use pulse directly, also pulseaudio steps away really transparently when some application tries to use plughw/hw device directly. Just be sure to get pulseaudio 0.9.8 or later, don't mess with older ones. -- Vladimir - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2008. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] FC8, no ALSA sound support
Hi Gianluca Cecchi! On 2007.12.31 at 14:07:51 +0100, Gianluca Cecchi wrote next: Since old days I've been using 9 and 0 keys to decrease/increase the app volume of mplayer. ( / and * keys, with their placement on numerical pad I suppose, should do the same) And when you restart it, it uses the latest one set... perhaps Or I didn't understand your comment at all This only works if your sound card supports hardware mixer. A lot of modern cards, e.g. usb ones don't have one. With pulseaudio, it doesn't matter whether one has it; also the volume level you've set is per-application. I.e. with pulseaudio it works exactly as you described, BUT that volume doesn't affect other applications. Without PA: you stop playback in your audacious (100% volume). You run mplayer. You set mplayer's volume level to 50%. You run mplayer again. Your volume level is still 50%. You resume playback in audacious. Ack! your volume is 50% even there. With PA: everything is the same, except your volume level in audacious is still 100%. And if you run mplayer again, the level would be 50% in it (of course, you can disable PA feature which remembers volume level upon application exit) -- Vladimir - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2005. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] FC8, no ALSA sound support
Hi Peteris Krisjanis! On 2007.12.31 at 15:27:54 +0200, Peteris Krisjanis wrote next: If mplayer (not gmplayer) has a volume control I've yet to find it. To my mind, having all the volume controls in one place rather than in each app makes much more sense. But then I have a mixer :-) There's now a master fader (pulseaudio) and per-app faders which just seems to make sense because it has a good hardware analogue with the mixer. I admit if this is done right, with good GUI and understandable functionality, would be killer feature. For now, it is confusing, but let's hope it'll improve. It's done pretty good. Applications don't know about per-application volume levels, for each of them just main volume exists, but from PA perspective, each has its own volume control. And pavucontrol application to control each app's mixer is pretty good. It has simple interface, too. First part is nice feature, but overestimated - I would definitely stop and relaunch movie when changing sound outputs - because then I have to change wires, etc. there is no fun of going on stuff while you It's quite useful with usb audio devices (soundcards/speakers/headphones). Also, I use it to redirect sound from my notebook speakers to audio system connected to desktop PC. Unfortunately, it works reliable only with ethernet connection, but maybe my home wifi just sucks. Have you used PA? It doesn't block the sound card, or at least mine doesn't. And the whole point of PA, at least in Fedora 8, is that for most people it does just work. Ok, this thread started as people claiming that PA blocked their sound card. Obviously, I voiced in just because I have exactly the same experience - PA blocks everything, even mixer. The thing is, usage of PA still allows direct access to alsa/oss devices. No more killall esd, killall jack and so on hacks, because even when PA daemon is running, it closes all sound devices as soon as its latests client exits. Of course it blocks sound cards when someone is playing something though pulseaudio, but that's not the point (btw PA allows you to forcefully disconnect clients, though a nice gui, of course). But the fact that having pulseaudio running doesn't automatically make your sound devices blocked really makes a difference. Ok, sorry, I was using Debian meaning for unstable, which means kinda big number of complains, doesn't work all the time, etc. Use pulseaudio 0.9.8 and you'll (probably) be fine. Don't toy with earlier versions, you don't want any more bad experience, right? -- Vladimir - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2005. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] M-Audio Audiophile USB and digital out
Hi Stephan Seitz! On 2007.12.16 at 17:08:25 +0100, Stephan Seitz wrote next: with too much apps, so better don't use it at all, and better yet, run jack or pulseaudio sound server (latter is more compatible one with general desktop setup, but you'd better be using very recent version of it) and don't use hw/plughw devices directly in other apps. I tried Pulseaudio server 0.9.5 (Debian Lenny) configuring it with Like I said, you shouldn't use such old version of pulseaudio. The first actually usable for general desktop setup is heavily patched 0.9.7, as it appeared in fedora 8; now 0.9.8 is available and includes all fixes in main tree. „load-module module-alsa-sink device=plughw:0,1 sink_name=output” and This is fine but not really required since pulseaudio should detect and use correct audio device by itself. Or at least you can switch it manually after startup with pavucontrol. If someone can tell me how to configure Pulseaudio I will try it. Install recent version and all the useful utilites and you'll be fine. Also it's a good idea to lock sampling frequency to 48k or 96k and increase resampling quality (resample-method = src-sinc-medium-quality), though some people may prefer no resampling at all (however things will become worse when mixing is at work then). -- Vladimir - SF.Net email is sponsored by: Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://ad.doubleclick.net/clk;164216239;13503038;w?http://sf.net/marketplace ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] M-Audio Audiophile USB and digital out
Hi Stephan Seitz! On 2007.12.15 at 20:53:57 +0100, Stephan Seitz wrote next: I’m got a M-Audio Audiophile USB soundcard and I’m trying to get it work under Linux. I’m using kernel 2.6.24-rc4 with alsa version 1.0.15. Since the soundcard will be used for digital out, I tried to play something using the PCM device iec958 but I got no sound at the receiver. I was using different options for device_setup, but I think 0x09 should be the right choice (I’m playing 16bit flac files, 44 or 48kHz). Can anyone help me configuring this soundcard? Thanks in advance! Actualy, there is documentation you should read when using that card. You probably can find your in kernel-doc package on your system (file /usr/share/doc/kernel-doc*/Documentation/sound/alsa/Audiophile-Usb.txt or something like that), or you can read it online on http://www.mjmwired.net/kernel/Documentation/sound/alsa/Audiophile-Usb.txt To sum it up, try using plughw:0,1 device for playback. hw won't work with too much apps, so better don't use it at all, and better yet, run jack or pulseaudio sound server (latter is more compatible one with general desktop setup, but you'd better be using very recent version of it) and don't use hw/plughw devices directly in other apps. -- Vladimir - SF.Net email is sponsored by: Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://ad.doubleclick.net/clk;164216239;13503038;w?http://sf.net/marketplace ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] best card for bitperfect SPDIF I/O?with?external clock sync ?
Hi Bill Unruh! On 2007.11.24 at 10:11:16 -0800, Bill Unruh wrote next: Thank you for all your responses! That is quite normal It is in fact good hearing (although many kids can hear up to 22-25 kHz). And it will get worse, especially if you like That's the main reason why I asked - I heard that people can hear 22khz or about that when they are young, than listening gets worse and they don't hear very high frequencies anymore - and since I consider myself still young ;), I was kind of disappointed by the fact that I can't hear not only 22khz, but even 20khz, and when I discovered that I can't hear even 18khz, I was kind if scared - is my hearing going down due to headphone usage? Thanks for clarifying this issue. listening to music on your headphones. Almost all headphone users have their headphones cranked up WAY to loud, and that destroys the nerve cells in the ear. A bus, going up a hill, has sound levels inside of the order Hey hey, I know what's good for me ;) I only listen to headphones in quiet places nowadays and at comfortable volume levels. of 80dB and in order to hear the music people crank up their heaphones to 90 or 100 dB. After only a few years of that your threshold will be down to Actually there are solutions, like good in-ear noise isolation headphones, for example ER6 and ER4 are pretty good (though expensive) - http://www.etymotic.com/ephp/er6.aspx, with background isolation over -30dB. Actually I thought about using such thing when I'm outside, the only problem is that I think it may be too dangerous to walk in the city and cross the streets with such isolation. There are also active noise-cancelling headphones, though I'm not too fond of them. 14kHz then 8kHz then 3kHz. With any luck you will effectively be deaf by the time you are 40, and can join the ranks of almost all rock musicians. It can't be that scary. You mean that all people who are listening to headphones in bus are going to end like this? We'll become a deaf nation then. Nature must have thought of something to prevent this from happening. Or, science will help ;) -- Vladimir - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2005. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Bitperfect Ears
Hi Rene Herman! On 2007.11.25 at 15:22:59 +0100, Rene Herman wrote next: That 16 kHz -60dB is just about my threshold with good headphones, good card set to 0 dB and external amplication cranked up. -57 I hear always, at -60 it's a little flaky. At those levels, 12-0dB actively hurts... Well it's not like it means a lot, without knowing resulting loudness of the signal - for example I can barely hear that -57 at half volume level of my headphone amplifier, but music is too loud to unbearable at that level, and comfortable level for replaygain-corrected music is about quarter volume. However, I have no idea about actual volume of that sound.. No way I would try to playback 12-0dB sample at full volume, since both 12-9dB and 16-9dB generate very unpleasant and loud signal at my usual quarter volume level. If I were to playback 20khz at levels over 100 dB and still wouldn't be able to hear it, would it mean that it's the same as if I don't hear anything at all? Sound still can damage ears even though you don't hear it, IIRC. PS 12-3dB from headphones at quarter volume level can be heard clearly from 5m away! On the other hand, 12-0dB sample seems to be either broken, or my system can't playback it - I head very high lower-frequency noise when playing it (resampling problem, maybe?), of course at good range away from headphones. PPS my head hurts :( Enough with experiments for today. I don't recommend anyone to toy with these 0dB samples.. -- Vladimir - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2005. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] best card for bitperfect SPDIF I/O?with?external clock sync ?
Hi Gene Heskett! On 2007.11.23 at 11:56:56 -0500, Gene Heskett wrote next: Anyway, in analog mode, are you sure there is no option to switch off bandwidth filter? No one in their right mind would want to do that as the aliasing would drive you up a wall. The other delay distortions the filter might give are 100's of times more tolerable to listen to. Er.. I meant in 96khz mode. Or in 48khz. I understood original letter as if the card has that bandwidth limitation on any sampling frequency, and this is a problem. Of course at 44.1khz D-A conversion it's desired to have something like that. Not that it matters in digital mode, since it's up to DAC filters. PS a bit of OT: I'm 24, and I barely hear 18khz (in headphones), unless it's VERY loud - I can hear only up to 17500-17800 clearly at average volume level. Is there something wrong with my ears or it should be like this? -- Vladimir - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2005. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] best card for bitperfect SPDIF I/O with external clock sync ?
Hi Darrell Bellerive! On 2007.11.22 at 17:54:19 -0800, Darrell Bellerive wrote next: I have never been happy with this card. While it works okay for playing basic sound, getting it to do anything more sophisticated is pure black magic. For example, I have never gotten full duplex to work. Well now, when pulseaudio era came, hopefully it's not true anymore. Also while the Audiophile 24/96 does sample at 96 KHz, the audio bandwidth of the card is limited to 22 Hz to 22 KHz +/- 0.4 dB. How does that matter when all is needed is digital output? Anyway, in analog mode, are you sure there is no option to switch off bandwidth filter? -- Vladimir - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2005. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] M-Audio Audiophile 2496 was: Re: best card for bitperfect SPDIF I/O with?external clock sync ?
Hi Darrell Bellerive! On 2007.11.23 at 04:25:56 -0800, Darrell Bellerive wrote next: Well now, when pulseaudio era came, hopefully it's not true anymore. Pulseaudio shows some promise, but not all apps support it yet. Wasn't Most of them do - some through wrappers or alsa-pulse routing, for example SDL, openal and wine, but unlike jack, it works much better. The most problematic thing is gstreamer, it works with pulse, but audio quality suffers for unknow reason (not much, but quite noticeable on some material, and very annoying if you have good ears). Besides, with stuff like per-application mixers which stores its state (a feature you'll fall in love with instantly), on the fly audio card detection and stream switching, network transparency with transparent stream migration (just a few clicks to move sounds from your notebook to home audio system connected to main pc or home media server when you're home - I find it really useful), high quality resampling and so on really makes it the most desired think to have in the core of your linux audio system. JACK, and then GStreamer, and soon Phonon, supposed to solve all our Jack has completely different purpose. It is good (bwt you can run pulse on top of jack, and some day probably will be able to run jack on top of pulse - though it wouldn't have much sense). gstreamer and photon are systems working at different level. They don't conflict with any sound server and support most of them. audio problems? Yet another sound server. You might think of it like this, however after discovering pulseaudio I think that it's holy grail of linux audio we've been searching for years. Still a bit edgy, though. Also while the Audiophile 24/96 does sample at 96 KHz, the audio bandwidth of the card is limited to 22 Hz to 22 KHz +/- 0.4 dB. Anyway, in analog mode, are you sure there is no option to switch off bandwidth filter? If there is, it is not documented. http://alsa.opensrc.org/index.php/Ice1712 http://alsa.opensrc.org/index.php/Envy24control Well there are Audiophile 192, Audiophile USB and other cards in Audiophile series - most are even more interesting than older Audiophile 24/96. So I don't really see any problem. -- Vladimir - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2005. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] best card for bitperfect SPDIF I/O with external clock sync ?
Hi Sergei Steshenko! On 2007.11.21 at 23:24:35 +0200, Sergei Steshenko wrote next: Regarding soundcard and syncrhonization - M-Audio Revolution 7.1, and quite possibly M-Audio Revolution allow you to use external clock source. I meant M-Audio Revolution 7.1, and quite possibly M-Audio Revolution 5.1 allow you to use external clock source. In theory, yes. In practice, I wasn't able to make my M-Audio Audiophile USB get clock from external source. Well it kinda works, but at some point distortions appear, and one must force clock resync or something like that by turning card off and on. I wanted to create setup similar to this, and one of the things I learned that in order to reduce jitter, you'd want to have power as clean as possible. On-board soundchips produce lowest quality signal, pci/pcie boards have much better filtering and produce better signal, but if you want something better, you have to use card which isn't powered by PSU of your PC, and doesn't suffer from problems of its signal. So if you want best digital audio, you probably should look among external cards (usb/firewire) which aren't bus powered, and use external AC adapter. Interface doesn't matter as long as card doesn't get power from it, so choose most compatible card. I picked Audiophile USB, which supports up to 24bit/96khz (though most likely you'll use it in 24bit/48khz mode). It also supports almost any sound rate without resampling, i.e. you can drive your DAC at 44100 in bit-exact mode and get highest quality possible signal, and resampling would happen only in DAC or won't happen at all (internal upsampling mode is recommended for most modern DACs though). As about quality, all I can say is that digital output from this card sounds much better than digital output from Audigy 2 ZS (both in regular or through p16v path, presumable working in bit-exact mode in the last case). As about external clock.. Audiophile usb _supports_ syncing from spdif in, but when I tried to connecting spdif out of my DAC to spdif in on card and enabled that spdif in order to synchronize, I got just noise (or very distored signal) from spdif out on card. I could get it working when spdif out on dac was set to no signal or signal from some of the unsed inputs, at some conditions I was even able to get it working with spdif out set to output signal from spdif in to which sound card was connected, but this configuration wasn't very stable and after a few hours of usage the distortions could start again. Switching the source of spdif out when it was used as clock source for the card also could produce some very strange results. Due to strangeness of this recursive scheme (imagine signal path: card output - dac input - dac output - card input - clocks from CARD's original output are used to drive next card output?) and me not understanding how and at what point re-clockings occur, but most important: completely failing to hear any advantages of this setup comparing to simplest path, I stopped experimenting with it. Now I just enjoy music and I'm quite satisfied with it ;) -- Vladimir - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2005. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] hw or plughw
Hi paul blakeley! On 2007.11.20 at 12:26:29 +, paul blakeley wrote next: Can someone please explain the differences between these? What impact they have on the application? plughw supports much more sample formats / channel configurations than underlying hardware supports natively, and performs conversion if needed. hw performs no conversion, but supports less configuration, sometimes only very obscure ones, but when used you can rest assured that no conversion takes place. Mostly you'd want these conversions to take place, like mono-stereo conversion or S16LE-S32LE conversion etc (all depending on your hardware) If I need to drive the sound card directly should I use 'hw'? You can use hw, but it isn't really recommended unless you REALLY know how to use it and have support of every weird format in all possible combinations. That's about underlying hardware details, and most applications don't want to deal with them. For example my soundcard supports only S24_3BE format; you can't open hw device in any other mode, if you want to output S16LE (most applications never heard of S24_3 formats, let alone BE variations), you must use plughw, there is no other choice. Or p16v device on audigy2 supports only 8-channel modes; you can't output stereo signal to it, no matter how you try. So you either can output 8 channels to hw device or let plughw to do stereo-8ch conversion for you automatically. Unless you want to take care of all these little details, using hw is probably not a good idea. Though it's required for some applications because you don't actually know if/what kind of conversion takes place when you use plughw, most application would trouble users much less if they were to use plughw instead of hw. Actually, almost all application shouldn't even use plughw, sticking to default device, to allow software mixing, jack/pulse routing plugins, user choosen conversions to take place. If you do anything else, you create problems for users, so you must have really good reasons to do so.. -- Vladimir - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2005. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Problems getting alsa-jack plugin to work
Hi Robert Gruendler! On 2007.10.03 at 19:14:05 +0200, Robert Gruendler wrote next: Anyone knows what i'm doing wrong here ? Yup, you are using alsa-lib 1.0.14 which is known to be broken. You have either to downgrade to alsa-lib 1.0.14rc3 (last working version), or to upgrade to 1.0.15rc3 (latest release) - pick your choice.. For example, in fedora 7 updates alsa-lib is broken, but you can downgrade to original fedora 7 alsa-lib 1.0.14-0.4.rc3.fc7 packages (using rpm -F --oldpackage) to make alsa-jack plugin work (it's the easiest path, and nothing'll break). Note that this problem is strictly alsa-lib related! If you choose to download and compile latest alsa packages yourself, don't touch alsa-drivers, the kernel part - 1.0.14 is perfectly fine. -- Vladimir - This SF.net email is sponsored by: Splunk Inc. Still grepping through log files to find problems? Stop. Now Search log events and configuration files using AJAX and a browser. Download your FREE copy of Splunk now http://get.splunk.com/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] How can I setup the alsa jack plugin?
Hi [EMAIL PROTECTED]
On 2007.06.17 at 18:51:24 +0200, [EMAIL PROTECTED] wrote next:
does anyone of you know how to setup the alsa jack plugin? Jackd is working
flawlessly for me, but the alsa jack plugin isn't. I used the following
asound.conf:
pcm.!default {
type plug
slave {
pcm jack
}
}
Try together with rate plugin. If your jack is configured to sample rate
other than 48000, change accordingly. Here is my configuration (my_jack
is explict device which uses jack, I forgot details, but I had to use
more complex configuration for !default, with most of it taken from
default alsa config file - YMMW).
I forgot why I added plug layer between rate and jack ones, but removing
it breaks something.
pcm.my_jack {
type rate
slave {
pcm jackplug
rate 48000
}
converter samplerate
}
pcm.jackplug {
type plug
slave.pcm jack
}
pcm.jack {
type jack
playback_ports {
0 alsa_pcm:playback_1
1 alsa_pcm:playback_2
}
capture_ports {
0 alsa_pcm:capture_1
1 alsa_pcm:capture_2
}
}
pcm.!default {
@args [ CARD ]
@args.CARD {
type string
default {
@func getenv
vars [
ALSA_PCM_CARD
ALSA_CARD
]
default {
@func refer
name defaults.pcm.card
}
}
}
# use card-specific definition if exists
@func refer
name {
@func concat
strings [
cards.
{
@func card_driver
card $CARD
}
.pcm.default:CARD= $CARD
]
}
default {
# use jack output as default
type rate
slave {
pcm jackplug
rate 48000
}
converter samplerate
}
}
Note that most application will work correctly, however some won't (xine
usually works, but can have problems with it for example). Also if one
application that uses jack plugin pauses/locks up, you'll get problems
with all other applications using it (or maybe even with application
that directly use jack). For example, when some media player which uses
gstreamer pauses playback, mplayer locks up - kinda wrong example, since
both mplayer and gstreamer have jack output drivers, however I
experienced this problem until very recently, till gstreamer got working
jack output plugin.
If you want more perfect solution, look at oss2jack. It works better,
however there is no need to use it for all application. You should use
alsa solution, and if some application has problems with it, try
using oss2jack for it. Of course you'd better disable oss emulation
layer in alsa completely - don't load the modules.
--
Vladimir
-
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Re: [Alsa-user] Recommendation for high end hardware mixing PC soundcard?
Hi Arthur Marsh! On 2007.02.18 at 22:44:29 +1030, Arthur Marsh wrote next: As previously discussed here, I'm tending towards an Audigy 4 Pro while I can still get one in Australia - I don't have the free PCI slots or fast enough PC to use another solution whilst keeping hardware synth and mixing. I wonder why are you saying this. First, I don't understand about PCI slot - Audigy requires one, you should go to USB cards if you don't have a free one. Second, why do you think that mixing streams in software takes that much of CPU power? I stopped jack, restarted it with different options and started playing 20 audio tracks via alsa-jack chain with default alsa 44.1-48khz sample rate conversion. About 30 seconds after that, I ran ps (top shows cpu usage for last few seconds, while only averaged cpu usage since start shows meaningful results). mosgalin 6160 0.1 1.2 40640 26552 pts/9SLl 15:38 0:00 jackd -t1000 -R -P89 -dalsa -Phw:1,1 -r48000 -p512 -n2 -H mosgalin 6184 0.2 2.9 74084 59768 pts/4SLl 15:38 0:00 aplay -D jackplug track01.wav mosgalin 6186 0.3 2.9 74084 59768 pts/4SLl 15:38 0:00 aplay -D jackplug track02.wav mosgalin 6188 0.3 2.9 74084 59768 pts/4SLl 15:38 0:00 aplay -D jackplug track03.wav mosgalin 6191 0.3 2.9 74088 59772 pts/4SLl 15:38 0:00 aplay -D jackplug track04.wav mosgalin 6193 0.2 2.9 74084 59768 pts/4SLl 15:38 0:00 aplay -D jackplug track05.wav mosgalin 6195 0.3 2.9 74088 59772 pts/4SLl 15:38 0:00 aplay -D jackplug track06.wav mosgalin 6197 0.2 2.9 74084 59768 pts/4SLl 15:38 0:00 aplay -D jackplug track07.wav mosgalin 6199 0.3 2.9 74088 59772 pts/4SLl 15:38 0:00 aplay -D jackplug track08.wav mosgalin 6203 0.3 2.9 74088 59772 pts/4SLl 15:38 0:00 aplay -D jackplug track09.wav mosgalin 6205 0.3 2.9 74084 59768 pts/4SLl 15:38 0:00 aplay -D jackplug track10.wav mosgalin 6207 0.2 2.9 74084 59768 pts/4SLl 15:38 0:00 aplay -D jackplug track10.wav mosgalin 6209 0.2 2.9 74088 59772 pts/4SLl 15:38 0:00 aplay -D jackplug track09.wav mosgalin 6211 0.2 2.9 74084 59768 pts/4SLl 15:38 0:00 aplay -D jackplug track08.wav mosgalin 6213 0.3 2.9 74088 59772 pts/4SLl 15:38 0:00 aplay -D jackplug track07.wav mosgalin 6215 0.2 2.9 74084 59768 pts/4SLl 15:38 0:00 aplay -D jackplug track06.wav mosgalin 6217 0.3 2.9 74084 59768 pts/4SLl 15:38 0:00 aplay -D jackplug track05.wav mosgalin 6219 0.2 2.9 74088 59772 pts/4SLl 15:38 0:00 aplay -D jackplug track04.wav mosgalin 6222 0.2 2.9 74084 59768 pts/4SLl 15:38 0:00 aplay -D jackplug track03.wav mosgalin 6224 0.3 2.9 74084 59768 pts/4SLl 15:38 0:00 aplay -D jackplug track02.wav mosgalin 6226 0.2 2.9 74088 59772 pts/4SLl 15:38 0:00 aplay -D jackplug track01.wav mosgalin 6236 0.0 0.0 62316 788 pts/3S+ 15:39 0:00 grep jack So I don't know where you got the assumption that mixing uses a lot of CPU power. Unless by not fast enough PC you mean 486 ;), it shouldn't burden your PC that much. As about software synth, well, first of all, hardware synth on Live/Audigy really sucks. It worked nicely for some very old games ;), but either in general, or on linux with alsa drivers it produces output much worse than timidity. So I doubt you'll be able to do anything useful with it anyway. Also, about 10 years ago, when I had P166MMX system, it was mostly enough for software synth - only a few midi files with a lot of tracks required more powerful system. Back than, hardware synth on Live was useful. However, I fail to understand why are you trying to re-create that kind of configuration now. -- Vladimir - Take Surveys. Earn Cash. Influence the Future of IT Join SourceForge.net's Techsay panel and you'll get the chance to share your opinions on IT business topics through brief surveys-and earn cash http://www.techsay.com/default.php?page=join.phpp=sourceforgeCID=DEVDEV ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Recommendation for high end hardware mixing PC soundcard?
Hi Peter! On 2007.02.18 at 16:13:46 +0100, Peter wrote next: The problem is, I use lots of different software that plays sounds, including software that only supports OSS (e.g. all old closed software like the loki games, TeamSpeak 2), and I have never managed to get my linux box to just allow running these apps, and some music etc. all at once. This is a linux problem, when using windows software mixing just works, and nobody cares (or even knows) if they own soundcards with hardware mixing capabilities. Would the above be possible with jack? Or is my software use-case just far from normal and nobody cares? No, jack isn't the solution that increases compatibility. Actually, it decreases it, since only apps that support jack directly work perfectly, the one that support only alsa /can/ be redirected to jack via alsa-jack wrapper, but it works only as long as they all behave good, once one of them exhibits any problem, all your music stalls (fortunately, since most of programs support either jack directly, or work though jack-enabled backend like gstreamer, pulseaudio, portaudio, this problem is quite rare). The purpose of jack is completely different. You can't use oss programs with jack, however you might get some luck with aoss wrapper. oss-aoss-alsa-jack is a horrible chain, and it doesn't work for me - YMMW. On the other hand, there is direct oss-jack wrapper: http://fort.xdas.com/~kor/oss2jack/, but I never tried it. When I started using jack, I suddently discovered that I don't have any oss-only application on my desktop; all the programs I ever used support alsa, or even jack. So don't blame linux or alsa, blame oss! Well, don't blame it if you use oss drivers, but if you using alsa, stay away from oss emulation and you'll be fine. As about your problem - believe it or not, it is the same in linux now! Most distros automatically configure dmix for you, and you shouldn't care how much hardware streams your card supports. Of course, all your apps must be alsa-enabled, but I don't think most people ever heard of oss-only apps nowadays. I don't know anything about teamspeak, but I heard that name a lot in jack or alsa context, so there should be some solution - try alsa wiki and google. As about games, well - wine supports jack output. Never actually tried using it, but it should work, I guess. Common wine alternative, crossover from codeweavers also has jack plugin. Not sure about the last alternative, cedega from transgaming, but probably it supports jack too.. Can't say anything about loki games, but.. aren't they VERY old? Do they ever work? I have Loki Demo CD with a few game demos on it, but it's about 7 years old, I really doubt it ever works with modern glibc/other components, not to mention x86-64 systems (and I don't have any legacy 32-bit x86 desktop around me to try it), and I heard that loki is out of business for a very long time. I won't argue with you that if you really need to run old loki games, there isn't any better solution that using creative card. On the other hand, I can't think of any other reason to use them, there are much better alternatives. -- Vladimir - Take Surveys. Earn Cash. Influence the Future of IT Join SourceForge.net's Techsay panel and you'll get the chance to share your opinions on IT business topics through brief surveys-and earn cash http://www.techsay.com/default.php?page=join.phpp=sourceforgeCID=DEVDEV ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Recommendation for high end hardware mixing PC soundcard?
Hi Peter! On 2007.02.17 at 22:45:32 +0100, Peter wrote next: 0) linux people don't care about hardware mixing [why? software mixing in linux is a big building site and was never capable to provide me the comforts and possibilities I got with hardware mixing on the emu10k1 based cards.] The truth is, hardware mixing isn't the hot thing right now. Just like hardware MIDI synth, it's more or less deprecated. Low-end (like integrated ones) sound cards don't support it because they must be cheap. High-end professional cards for musicians don't need it because it's better to have practically unlimited amount of streams that you can mix in software with 32-bit precision on any sampling rate than to be limited in number of streams, sampling rate and mixing/resampling conversion quality. This doesn't applies to all high-end devices though, software processing is preferred now, still there is some use for hardware processors, but these devices are rare and expensive. As about middle-ranged cards, they are either about delivering higher quality sound than onboards cards (models from M-Audio, Terratec) - they don't need hardware mixing because software works very well nowadays, or about gaming (Creative cards) - the only ones that truly need hardware mixing, to unload mixing big amount of streams off processor. In other words, if you really desire hardware mixing, buy creative card. Audigy 2, Audigy 4, then X-Fi.. well you know where it leads. So I suggest you to rethink, maybe you don't need hardware mixing that much after all. As a former Live! 5.1 and Audigy 2 ZS user that now uses M-Audio Audiophile USB, I can tell you that problems with software mixing in linux aren't that big. If you don't need exceptional sound quality, just use onboard card; if you do, buying Creative card is wrong. Just buy something like M-Audio Revolution, Terratec Aureon, or more expensive ESI Juli@, EMU 1212M instead. Or, better yet, some USB card that feeds off AC adapter, not USB port - believe me, you won't regret it. I still have Audigy 2 ZS plugged in, but there is no way I'll use it for playing back any music - external M-Audio card is so much better when it comes to audio quality, both analog and digital outputs are much superior (also integrated headphone amplifier on this model should be mentioned - not perfect, but better than cheap OP-AMP models like ART HeadAMP). I use jack as sound server. mplayer supports jack, recent gstreamer finally has working jackaudiosink, other application either use jack directly or work via alsa-jack redirection plugin. If you distro doesn't support jack it might be a little pain to rebuild required pieces of software with jack support, but it's not that hard, and some modern distros have very good jack support. Of course, jack isn't the only solution, but it's the best one if you want to be sure you are not compromising any quality. -- Vladimir - Take Surveys. Earn Cash. Influence the Future of IT Join SourceForge.net's Techsay panel and you'll get the chance to share your opinions on IT business topics through brief surveys-and earn cash http://www.techsay.com/default.php?page=join.phpp=sourceforgeCID=DEVDEV ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] USB and 8000 Hz
Hi Michael Bourgeous! On 2007.02.13 at 15:14:40 -0700, Michael Bourgeous wrote next: Hi everyone, I have managed to get my USB soundcard to work fine with alsa-utils but unfortunately it is only possible to open it in 44100 and 48000 Hz. I wan't to be able to open it in 8000 Hz (natively), it would be perfect if it supports both 8000 and 16000 natively. I know how to switch the rate in asound.conf, but that wouldn't be native. Does anyone now of an USB soundcard that supports 8000 Hz natively? To answer original post: my M-Audio Audiophile USB seems to support that output frequency, as well as any other (16000, 24000, 11050 etc). I.e. it is possible to open hw device with that frequency, for example by running jack with -r8000 parameter and it produces correct sound on analog output. However, 24000 seems to be lowest frequency that works nicely on digital output; 16000 still works, but noticeable background noise appears, and no sound at all is produced at 12000 and lower frequencies. Though this may be my DAC limitation as well. Note that Audiophile USB creates two devices that can be used simultaneously, one for analog output, another one for digital. -- Vladimir - Using Tomcat but need to do more? Need to support web services, security? Get stuff done quickly with pre-integrated technology to make your job easier. Download IBM WebSphere Application Server v.1.0.1 based on Apache Geronimo http://sel.as-us.falkag.net/sel?cmd=lnkkid=120709bid=263057dat=121642 ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] 5 meter USB cable - Hearing ticks
Hi Jean-Michel Pouré! On 2007.02.09 at 12:04:53 +0100, Jean-Michel Pouré wrote next: I have two USB 2.0 Aureon MK2 devices, keyboard and mouse connecting to an USB 2.0 480 Mb/s hug. When plugin a one meter armored cable, the audio devices work perfectly. When connecting with a five meter non-armoered cable, I can hear clicks during play. I don't think you can do anything about that. 5m is too far for stable USB 2.0 connection and also too far for many 1.1 devices. USB and long cables are no-go; though you can buy even longer cables, very few low-speed devices work stable over them. I've seen numerous issues with long USB cables; for example, an HP printer which wasn't stable with 3m cable - computer was even losing the device sometimes. Audio cards usually don't need high speeds and USB 2.0, but they should be quite sensitive to cable problems. There are some possible solutions - you can try to manage with shorter cable, search for highest-quality cable, buy a good _active_ (powered, cheap ones won't do) hub and use two shorter cables. Of course, there are other, more expensive solutions, like using USB-over-UTP transmitter, they cost over $50, but increase range up to 30-50m... -- Vladimir - Using Tomcat but need to do more? Need to support web services, security? Get stuff done quickly with pre-integrated technology to make your job easier. Download IBM WebSphere Application Server v.1.0.1 based on Apache Geronimo http://sel.as-us.falkag.net/sel?cmd=lnkkid=120709bid=263057dat=121642 ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
[Alsa-user] Few questions regarding emu10k + p16v
Hi everybody. I have Audigy 2 ZS card and alsa driver 1.0.11 (from fc 2.6.16 kernel). I was trying to experiment with p16v, and noticed a few things. I'm using analog output. 1) Distiortion / clipping. Regular playback sound fine at master/pcm @ 100%. However, output from p16v sounds really distorted when its only mixer, HD Analog Front is at 100%. I read that 0dB is at 100% mixer values for all modern creative cards, is this true? Because according to my hearing, clipping disappears at about 80 mixer value. 78-82 sound the best. This isn't really a problem, just like to clarify this issue. 2) The correct way of using p16v is plughw 0:4 with padded 32-bit s32le 48/96/192 khz data, correct? What about hw 0:4, can it be used? When trying to use it in aplay/mplayer/gstreamer, it doesn't work good with 48khz data, and with 96/192 plays very short fragment of sound very fast. Can it ever be used directly, or absolutely all user application can use only plughw? 3) p16v can be opened at other rates like 44.1khz, but sound quality isn't good - no wonder. However, the following thing bugs me. from p16v.h, sorry for long lines: #define SRCSel 0x60 /* SRCSel. Default 0x4. Bypass P16V 0x14 */ /* [0] 0 = 10K2 audio, 1 = SRC48 mixer output. * [2] 0 = 10K2 audio, 1 = SRCMulti SPDIF mixer output. * [4] 0 = 10K2 audio, 1 = SRCMulti I2S mixer output. */ /* SRC48 converts samples rates 44.1, 48, 96, 192 to 48 khz. */ /* SRCMulti converts 48khz samples rates to 44.1, 48, 96, 192 to 48. */ /* SRC48 and SRCMULTI sample rate select and output select. */ #define SRCMULTI_ENABLE 0x6e /* SRCMulti input audio enable. Default 0x */ /* SRCMulti converts 48khz samples rates to 44.1, 48, 96, 192 to 48. */ /* [7:0] The corresponding P16V channel to SRCMulti_I2S enabled if == 1. * [15:8] The corresponding E10K2 channel to SRCMulti I2S enabled. * [23:16] The corresponding P16V channel to SRCMulti SPDIF enabled. * [31:24] The corresponding E10K2 channel to SRCMulti SPDIF enabled. */ Is it my understanding that what I send to p16v in 96khz is converted to 48khz after that by default? Can I2S output be used with rates other than 48khz? If yes, how do I do this? And what's this stuff about converting 48khz to 44.1? Can I tweak these settings in runtime? Actually, should I? I don't really need 96khz output right now, only 24-bit 48khz one. But according to this, if I try to send 48khz signal, it gets converted to 44.1 and then back to 48khz.. What's with all this stuff? I'm completely confused now. Why must it be that complex? -- Vladimir - Take Surveys. Earn Cash. Influence the Future of IT Join SourceForge.net's Techsay panel and you'll get the chance to share your opinions on IT business topics through brief surveys -- and earn cash http://www.techsay.com/default.php?page=join.phpp=sourceforgeCID=DEVDEV ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Routing the Emu10k1 wavetable to the rear speakers
Hi Brian Dunn! On 2006.02.06 at 11:27:10 -0600, Brian Dunn wrote next: I read somewhere that the rear output DACs on my SBLive Value are superior to the front ones, so my monitors are connected to the rear I'd say that if you definitely hear the difference, you should consider buying some other card ASAP. Preferable not the one from creative, they don't make good cards. Well, it's not that their cards are bad, but each one has some flaws which makes sound quality worse that other cards of the same price. They have some good cards in professional line, like 1212M, but it isn't supported by alsa yet. If you can't hear the difference.. why bother? out. I'm using qsynth with jack when i'm focusing on audio, but it seemslike a shame to have that wave table and whopping 13megs of DRAM just sitting there... It would be nice to asfxload some small GM soundfont so i can tinker with ditties without having to run jack ( witch is inconvinient when i'm compiling, or have 13 firefox windows and some open office going on ). It seems there should be some way to route the synth channel to the rear speakers... maybe with an .asoundrc file? has anybody done this? emu10k wavetable synth doesn't need any special routing; also front channel is routed to the rear output by default, there is surround mixer in its past. If you have already unmuted it, you should hear the same sound from front and rear output jacks. It doesn't matter whether you play pcm sound, use software synth or the hardware one. However, I suggest you to stay away from hardware synth on sb live, it sucks. There is something wrong with it, it sounds right in windows, but the sound is different with alsa drivers in linux. Try timidity; it can use your favorite soundfonts and with the right tweaks it can sound quite good. It is also more conservative about memory - it loads only samples which are actually used. You can use huge (100-200mb) soundfonts and don't think about memory issues. (actually, why would you care about them anyway? Linux has pretty good VM subsystem, if you don't use the part of memory that contains soundfonts for a while, it gets swapped out and memory for your precious firefox windows is freed) -- Vladimir --- This SF.net email is sponsored by: Splunk Inc. Do you grep through log files for problems? Stop! Download the new AJAX search engine that makes searching your log files as easy as surfing the web. DOWNLOAD SPLUNK! http://sel.as-us.falkag.net/sel?cmd=lnkkid=103432bid=230486dat=121642 ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
