Linux-audio-user September 2004

linux-audio-user@lists.linuxaudio.org

161 participants
177 discussions

[linux-audio-user] M-Audio Sonica/Ozone/MobilePre/Transit firmware loader now works

by Clemens Ladisch

Hi, it has been reported that the 2.6.8 kernel has fixed the handling of USB resets, so now it's possible to use a Linux firmware loader to get these devices to work. You can get the loader at <http://usb-midi-fw.sf.net/>. Best regards, Clemens

20 years, 3 months

Re: [linux-audio-user] Converting sample rate: failed...

by davidrclark＠earthlink.net

Hi Erik, As you can see, we made it back after chasing Ivan around. Luckily we saw nothing but a few drops of rain. I'm sure others here or close to those here must have had bad luck instead of good, and my sympathies go out to them. Thanks for your response. I applaud your enthusiasm, but it appears to me that you are laboring under some misconceptions and misunderstanding of what I have posted. Among the misconceptions is a serious one that should cause concern amongst those who use libsamplerate. ------------------------- Minor problems: A good portion of your recent post is aimed at something I never claimed. It appears to me that you are failing to distinguish between two cases: 1) Series in general; 2) Series which have been previously prepared, specifically frequency spectra which are band- limited and lowpass filtered. >From my earlier post on the current subject: "The reason is that I assume that the input is band-limited, and this is usually true for my own work. Not only is it band-limited, but usually also tapered in the frequency domain, i.e. already effectively lowpass filtered." >From your recent post: "Now I will admit that if the signal is already bandlimited filter many not be necessary, but that is a different matter all together." No, it is certainly not a "different matter altogether." Your quote was nearly all I was saying as you can easily see, except that in addition signals I use (as well as most that others use) are also normally effectively lowpass filtered. This is more restrictive than your assumption, so filtering is even less necessary than you admit it to be. Your elementary examples from undergrad engineering courses and attempt to rephrase what you claim I said don't apply to the situation I've described; they have nothing to do with what I've said and are true but irrelevant statements which cause me to wonder what it is that you are trying to prove; you may want to give some thought to that: What is it that you are actually trying to prove or demonstrate to everyone? As a very minor note, you are also attempting to answer an ontological question (that something which has no observable effect on the universe can nonetheless be said to exist) which has plagued philosophers for millenia by merely declaring your opinion as fact. Here is what I actually wrote: "If nothing was removed or even altered, then no filtering has actually occurred." Note that I am not so reckless as to claim anything regarding existence or nonexistence here. ------------------------- ** Serious misconception **: Two paragraphs from your recent post: "If you work out the mathematical expression for your frequency domain converter, you will find that there is a time domain expression that is mathematically identical and that the time domain expression is in part a FIR lowpass filter very much like my converter." (You mean your implementation of Smith's converter, don't you?) "The difference between the two would be that my version uses linear interpolation into a very large table to obtain the filter coefficients while yours are more exact. However mine provides a *measured* SNR of at least 97dB." This second paragraph is considerably flawed and reveals a serious misconception on your part. Although the *starting point for the derivation* of both methods is the same, the method you are using as it is actually implemented is a very localized method in which only few samples are used to construct the new values. This is due to the *severe* truncation of the series. The horrible effects of this severe truncation are mitigated through the application of a Kaiser window which effectively localizes the estimation even further as it improves the result substantially. The method I'm using remains a very much more global method in which each new value is constructed from hundreds of thousands to millions of samples. Given that audio I process taper off to zero at the beginning and end --- in addition to being band-limited and effectively lowpass filtered --- I could have written a global method wherein each sample is the result of calculations involving ALL other samples rather than one which contains windows with hundreds of thousands to millions of samples. This is a far cry from what you are doing. This is all BEFORE the linear interpolation step which you claim is the only difference between our methods, a step which is totally unnecessary by the way for most fixed-sample-rate conversions (for example: unnecessary for 96,000 to 44,100, for 48,000 to 44,100, and for 44,100 to either of the others) and further degrades the sample rate conversion unnecessarily for the most common use. (This was done by Smith to permit variable-sample-rate conversions, a useful goal to say the least.) Now as I've already said, the sinc-based sample-rate converter you've constructed from Smith's work seems to work well in the sample I've listened to. But knowing what I know about sinc-based methods based on my own implementations, Smith's published work, and the lack of necessity for such a method for fixed-rate conversions, not to mention the fact that linear interpolation is being used where it really isn't necessary, leads me to reject libsamplerate and anything related to it unless it's forced on me by someone else via a dependency. I rarely do variable-rate conversions, so I have very little need for sinc- based sample-rate converters. ------------------------- For "ordinary" users: For those of you who have read this far --- hopefully not many: PLEASE don't be alarmed. As I have repeatedly said, Erik's implementation of Smith's work seems to work well. The inaccuracies in that method for fixed rate conversions, including the additional inaccuracy of the unnecessary linear interpolation, seem to be virtually inaudible. The main advantage is that a single method can be used for both fixed- and variable-rate conversions, and this is indeed a *considerable* advantage for a lib-based implementation. We are all indebted to Erik for creating libsamplerate, and this includes me. ------------------------- For developers: Developers, however, should at least be aware of what is going on there. I would advise anyone who has a *critical* dependence on sample-rate conversions to carefully read Smith's notes and to make sure that they understand what is going on. They should also experiment with Smith's technique, including implentation of different windows to see how bad it can be if things go wrong. Erik's "measurements" that I've seen so far aren't very useful for these worst-case situations. Smith may have had something more useful such as a bound of some sort, but frankly I don't recall at this moment and I'm not motivated to check it out at this time (sorry!). I have too much to do following this trip, and this post has taken all my spare time. Sorry for the length, but I wanted to clear up some misconceptions about what I've posted as well as raise a flag (without causing some sort of fiasco) for those who may not have had time to study sinc-based methods. Regards to all, Dave.

20 years, 3 months

[linux-audio-user] how to record streaming audio?

by airplays55＠yahoo.com

I want to run a sound recorder like audacity to record what is coming thru my soundcard when playing streaming audio thru xmms. However audacity can't open the audio port /dev/dsp when streaming audio is playing. This is on knoppix 3.4 using the OSS driver. Would using ALSA make this possible? __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

20 years, 3 months

[linux-audio-user] ices-jack pico-howto

by Joern Nettingsmeier

hi everyone! in case you were wondering how to get ices-jack to stream your jack graphs out on the net, here's a quick howto: browse svn.xiph.org, get the following modules from /trunk: ao, vorbis, ogg, ogg2, theora, speex, vorbis-tools, ogg-tools (do this even if you have ogg packages from your distro installed, it won't do no harm and makes sure you've got the latest'n'greatest) there's nothing interesting to configure afaik, so you can compile them (in that order) without interaction: for i in ao vorbis ogg ogg2 theora speex vorbis-tools ogg-tools; do svn co http://svn.xiph.org/trunk/$i; cd $i; ./configure && make install ; cd .. ; done from icecast/branches/kh, check out libshout, icecast, ices again, not really anything to configure, so the for-loop can do the grunt work... now fire up icecast, fire up ices, connect it to your jack graph, and the fun starts. the default config files are extensively commented, but here's my config, in case you need some more inspiration: http://spunk.dnsalias.org/download/ices.xml http://spunk.dnsalias.org/download/icecast.xml (the source and server run on different hosts, and icecast runs chrooted and as user icecast) btw, a graph with an ogg edge between ices-jack and xmms-jack vertices makes a nice delay effect :) if you use feedback, there's interesting sound deterioration due to repeated ogg-encoding/decoding and noise buildup. here's me toying around with my bass and such a setup: http://spunk.dnsalias.org/download/netjam.ogg have fun jörn -- "Debugging is twice as hard as writing the code in the first place. Therefore, if you write the code as cleverly as possible, you are, by definition, not smart enough to debug it." - Brian W. Kernighan Jörn Nettingsmeier Lortzingstr. 11, 45128 Essen, Germany http://spunk.dnsalias.org (my server) http://www.linuxaudiodev.org (Linux Audio Developers)

20 years, 3 months

Re: [linux-audio-user] MIDI on alsa with SBLIve

by davidrclark＠earthlink.net

Chris Cannam posted: > Because sfArk and sfpack compress soundfonts much better than > zip/gzip/bzip2 can. This is absolutely correct. The reason is that text files repeat character sequences exactly whenever words and other character combinations are used over and over again. The "normal" compression methods build tables and utilize this precisely repetitive nature effectively. Audio files repeat character sequences only approximately, so are not recognized as being almost the same even though they sound the same, hence are regarded as new sequences. True binary files (e.g. stripped executables) look pretty much random, also, to one of these compression algorithms. The random appearance of audio files is one reason why MP3, sfark, etc. were developed. To distinguish between these latter techniques: As most people know by now, MP3 is a lossy technique which means that information is lost never to be seen again. sfark is not. These soundfont compression techniques are a compromise between loss of information and effective compression. Some out there may be using MP3's in place of such compression schemes as sfark's, believing that MP3's are as good or better due to the good compression ratios obtainable. This is at a fairly heavy cost. I would advise against it generally, which is the reason I'm spending time posting this. MP3 is fine for distribution over limited-resource channels, but not so fine for soundfonts/samples. Hope this helps someone out there. Now I really am going to try to get caught up.... Regards to everyone, Dave.

20 years, 3 months

[linux-audio-user] A question on Interrupt Pin routing

by Louis Lam

Hello everyone, Currently I have my soundcard on IRQ9. /proc/interrupt shows this: CPU0 0: 208542 XT-PIC timer 1: 3710 XT-PIC keyboard 2: 0 XT-PIC cascade 8: 4 XT-PIC rtc 9: 1458611 XT-PIC Ensoniq AudioPCI 10: 1446 XT-PIC eth0 12: 98030 XT-PIC PS/2 Mouse 14: 41633 XT-PIC ide0 15: 563 XT-PIC ide1 NMI: 0 LOC: 208492 ERR: 0 MIS: 0 When I inspected further using lspci, I noticed under "interrupts:" of the soundcard it says "pin A routed to IRQ 9". There is another device, the SMBus Controller which has "Interrupt: pin B routed to IRQ 9". Does it mean that this IRQ is still shared? What does pin A/pinB mean and if they both route to the same interrupt, will it cause problems when recording? Thanks For any clues or info regarding this, Louis. P.S The Full lspci output is as follow: 0000:00:00.0 Host bridge: Intel Corp. 82875P Memory Controller Hub (rev 02) Subsystem: Intel Corp. 82875P Memory Controller Hub Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- Status: Cap+ 66MHz- UDF- FastB2B+ ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort+ >SERR- <PERR- Latency: 0 Region 0: Memory at f8000000 (32-bit, prefetchable) [size=64M] Capabilities: [e4] #09 [2106] Capabilities: [a0] AGP version 3.0 Status: RQ=32 Iso- ArqSz=2 Cal=0 SBA+ ITACoh- GART64- HTrans- 64bit- FW+ AGP3- Rate=x1,x2,x4 Command: RQ=1 ArqSz=0 Cal=2 SBA+ AGP+ GART64- 64bit- FW- Rate=<none> 0000:00:01.0 PCI bridge: Intel Corp. 82875P Processor to AGP Controller (rev 02) (prog-if 00 [Normal decode]) Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- Status: Cap- 66MHz+ UDF- FastB2B+ ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- Latency: 32 Bus: primary=00, secondary=01, subordinate=01, sec-latency=32 I/O behind bridge: 0000f000-00000fff Memory behind bridge: fc900000-fe9fffff Prefetchable memory behind bridge: f0800000-f47fffff BridgeCtl: Parity- SERR- NoISA- VGA+ MAbort- >Reset- FastB2B- 0000:00:03.0 PCI bridge: Intel Corp. 82875P Processor to PCI to CSA Bridge (rev 02) (prog-if 00 [Normal decode]) Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- Status: Cap- 66MHz+ UDF- FastB2B+ ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- Latency: 32 Bus: primary=00, secondary=02, subordinate=02, sec-latency=0 I/O behind bridge: 0000b000-0000bfff Memory behind bridge: fea00000-feafffff Prefetchable memory behind bridge: fff00000-000fffff BridgeCtl: Parity- SERR+ NoISA+ VGA- MAbort- >Reset- FastB2B- 0000:00:1e.0 PCI bridge: Intel Corp. 82801BA/CA/DB/EB/ER Hub interface to PCI Bridge (rev c2) (prog-if 00 [Normal decode]) Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- Status: Cap- 66MHz- UDF- FastB2B+ ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- Latency: 0 Bus: primary=00, secondary=03, subordinate=03, sec-latency=32 I/O behind bridge: 0000c000-0000cfff Memory behind bridge: fff00000-000fffff Prefetchable memory behind bridge: fff00000-000fffff BridgeCtl: Parity- SERR+ NoISA+ VGA- MAbort- >Reset- FastB2B- 0000:00:1f.0 ISA bridge: Intel Corp. 82801EB/ER (ICH5/ICH5R) LPC Bridge (rev 02) Control: I/O+ Mem+ BusMaster+ SpecCycle+ MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- Status: Cap- 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- Latency: 0 0000:00:1f.1 IDE interface: Intel Corp. 82801EB/ER (ICH5/ICH5R) Ultra ATA 100 Storage Controller (rev 02) (prog-if 8a [Master SecP PriP]) Subsystem: Intel Corp.: Unknown device 425a Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- Status: Cap- 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- Latency: 0 Interrupt: pin A routed to IRQ 10 Region 0: I/O ports at <unassigned> Region 1: I/O ports at <unassigned> Region 2: I/O ports at <unassigned> Region 3: I/O ports at <unassigned> Region 4: I/O ports at ffa0 [size=16] Region 5: Memory at 3fe3f400 (32-bit, non-prefetchable) [size=1K] 0000:00:1f.2 IDE interface: Intel Corp. 82801EB (ICH5) Serial ATA 150 Storage Controller (rev 02) (prog-if 8f [Master SecP SecO PriP PriO]) Subsystem: Intel Corp.: Unknown device 425a Control: I/O+ Mem- BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- Status: Cap- 66MHz+ UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- Latency: 0 Interrupt: pin A routed to IRQ 10 Region 0: I/O ports at ec00 [size=8] Region 1: I/O ports at e800 [size=4] Region 2: I/O ports at e400 [size=8] Region 3: I/O ports at e000 [size=4] Region 4: I/O ports at dc00 [size=16] 0000:00:1f.3 SMBus: Intel Corp. 82801EB/ER (ICH5/ICH5R) SMBus Controller (rev 02) Subsystem: Intel Corp.: Unknown device 425a Control: I/O+ Mem- BusMaster- SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- Status: Cap- 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- Interrupt: pin B routed to IRQ 9 Region 4: I/O ports at d800 [size=32] 0000:01:00.0 VGA compatible controller: nVidia Corporation NV5M64 [RIVA TNT2 Model 64/Model 64 Pro] (rev 11) (prog-if 00 [VGA]) Subsystem: CardExpert Technology: Unknown device 8888 Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- Status: Cap+ 66MHz+ UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- Latency: 32 (1250ns min, 250ns max) Interrupt: pin A routed to IRQ 11 Region 0: Memory at fd000000 (32-bit, non-prefetchable) [size=16M] Region 1: Memory at f2000000 (32-bit, prefetchable) [size=32M] Expansion ROM at fe9f0000 [disabled] [size=64K] Capabilities: [60] Power Management version 1 Flags: PMEClk- DSI- D1- D2- AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot-,D3cold-) Status: D0 PME-Enable- DSel=0 DScale=0 PME- Capabilities: [44] AGP version 2.0 Status: RQ=32 Iso- ArqSz=0 Cal=0 SBA- ITACoh- GART64- HTrans- 64bit- FW- AGP3- Rate=x1,x2,x4 Command: RQ=1 ArqSz=0 Cal=0 SBA- AGP- GART64- 64bit- FW- Rate=<none> 0000:02:01.0 Ethernet controller: Intel Corp. 82547EI Gigabit Ethernet Controller (LOM) Subsystem: Intel Corp.: Unknown device 3025 Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- Status: Cap+ 66MHz+ UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- Latency: 0 (63750ns min), Cache Line Size: 0x10 (64 bytes) Interrupt: pin A routed to IRQ 10 Region 0: Memory at feae0000 (32-bit, non-prefetchable) [size=128K] Region 2: I/O ports at bc00 [size=32] Capabilities: [dc] Power Management version 2 Flags: PMEClk- DSI+ D1- D2- AuxCurrent=0mA PME(D0+,D1-,D2-,D3hot+,D3cold+) Status: D0 PME-Enable- DSel=0 DScale=1 PME- 0000:03:02.0 Multimedia audio controller: Ensoniq ES1370 [AudioPCI] Subsystem: Unknown device 4942:4c4c Control: I/O+ Mem- BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- Status: Cap- 66MHz- UDF- FastB2B- ParErr- DEVSEL=slow >TAbort- <TAbort+ <MAbort+ >SERR- <PERR- Latency: 248 (3000ns min, 32000ns max) Interrupt: pin A routed to IRQ 9 Region 0: I/O ports at cc00 [size=64] ________________________________________________________________________ Yahoo! Messenger - Communicate instantly..."Ping" your friends today! Download Messenger Now http://uk.messenger.yahoo.com/download/index.html

20 years, 3 months

[linux-audio-user] Alsa problems with nforce2

by Leandro Monk

Hi: I 've try to use the alsa driver for my nforce2... when i try load the module snd-intel8x0, the system say me: /lib/modules/2.4.26-1-386/alsa/snd-intel8x0.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg /lib/modules/2.4.26-1-386/alsa/snd-intel8x0.o: insmod /lib/modules/2.4.26-1-386/alsa/snd-intel8x0.o failed /lib/modules/2.4.26-1-386/alsa/snd-intel8x0.o: insmod snd-intel8x0 failed My /proc/pci is: Multimedia audio controller: nVidia Corporation nForce2 AC97 Audio Controler (MCP) (rev 161). IRQ 5. Master Capable. No bursts. Min Gnt=2.Max Lat=5. I/O at 0xd800 [0xd8ff]. I/O at 0xdc00 [0xdc7f]. Non-prefetchable 32 bit memory at 0xec002000 [0xec002fff]. Bus 0, device 8, function 0: I use Debian sarge, and instal this packages: alsa-base, alsa-modules, alsa-utils. alsa-lib ¿Any idea to fix this problem? -- Leito Monk -------------------- Miembro de CaFeLUG www.cafelug.org.ar -------------------

20 years, 3 months

Re: [linux-audio-user] using cat to send a MIDI string ?

by Pedro Lopez-Cabanillas

On Wednesday 22 September 2004, Dave Phillips wrote: > I'm trying to send a MMC message to Ardour to set it up to receive MIDI > time code. Apparently Ardour will follow MTC but it has to receive an > MMC Start message first. So far I'm still unsuccessful, even with your > suggestions, but I'll keep at it. You can use "amidi", from "alsa-utils": $ amidi -S 'F0 43 10 4C 00 00 7E 00 F7' sends an XG Reset to the default port; `man amidi` for details. To can use "echo", too: $ echo -ne '\xf0\x7f\x01\x06\x02\xf7' > /dev/midi01 BTW, you can try a MIDI realtime message "start" like this: $ echo -ne '\xfa' > /dev/midi The message "f0 7f f7 06 02 f7" is not a valid MMC command, because the device-id 0xf7 is also the EOX status byte. It should be a number between 00 and 0x7f. Regards, Pedro

20 years, 3 months

[linux-audio-user] Turtle Beach Santa Cruz and MIDI

by Hans Fugal

How do I get my TB SantaCruz (cs46xx) to make sound when it gets midi events? I assume this would involve a wavetable, but I'd be happy with FM synth or any terrible sounding piano sound as well. My cpu is not really up to the task of sequencing and softsynthing at the same time. -- De gustibus non disputandum est.

20 years, 3 months

Re: [linux-audio-user] Turtle Beach Santa Cruz and MIDI

by Hans Fugal

On Tue, 21 Sep 2004 08:52:17 -0700, Russell Hanaghan <hanaghan(a)starband.net> wrote: > On Tuesday 21 September 2004 08:34 am, Hans Fugal wrote: > > The card works great for PCM. > > > > Which software I use to patch the midi depends on my mood, but usually > > aconnectgui or aconnect. > > > > Haven't been fiddling with jack or Qjackctl yet. > > > > I'm trying to get my midi controller keyboard to make noise out the > > soundcard, is all. > > Hmmm. Ok...so maybe the wrong midi device? How many midi devices show > in /dev/snd ? If more than one, try patching around. $ ls /dev/snd controlC0 midiC0D0 pcmC0D0p pcmC0D2p pcmC1D0c pcmC1D1c seq controlC1 pcmC0D0c pcmC0D1p pcmC0D3p pcmC1D0p pcmC1D1p timer The TB is card 0. > TB also has several outs for pcm (Surround stuff) but sound like your already > using the card for normal stereo audio. I will fiddle with the mixer more this evening. Perhaps the fm synth is on master1 or some other pcm... fugalh@falcon:~$ aconnect -oil client 0: 'System' [type=kernel] 0 'Timer ' 1 'Announce ' Connecting To: 63:0 client 64: 'CS46XX - Rawmidi 0' [type=kernel] 0 'CS46XX ' (the keyboard is turned off and I'm not at home to turn it on) -- De gustibus non disputandum est.

20 years, 3 months

← Newer
1
2
3
4
5
6
7
8
...
18
Older →

Jump to page:

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

Linux-audio-user September 2004