Linux-audio-user January 2005

linux-audio-user@lists.linuxaudio.org

207 participants
246 discussions

[linux-audio-user] Re: DeMuDi unusable on widescreen nVidia

by Jon B

> That FAQ is a little out of date. Neither I nor anyone else has had time to do > anything about it. Sorry. So you're saying it should be a wiki... > What you say is probably possible. Good luck! Possible, but not without a struggle. Does anyone know how to get nvidia drivers loaded for the 2.6.9-multimedia-686 kernel? I've tried 3 or 4 methods and haven't gotten it. Says something about VMALLOC error. I've made significant progress though. At this rate maybe I can restart all over again with just DeMuDi and get the video working from the start... Jon

20 years, 2 months

RE: [linux-audio-user] Switching from a MOTU828mk2 to RME Cards?

by Reuben Martin

This is quite a big leap your making. As much as I like to promote Linux based pro-audio, you should make very sure that changing out your entire system like that won't cause you problems. >I am gearing towards an RME Hammerfall card >and a behringer A8000 adat interface. I would rather recommend buying the Alesis AI-3 if you want to use the ADAT interface. This gives you more freedom to use various preamps. (IMO, berhringer's preamps suck) If you've got money you may want to invest in a nice preamp. http://www.soundpure.com/browseProductType.do?id=5 -Reuben

20 years, 2 months

[linux-audio-user] [OT] disk recovery problem

by David Cournapeau

Hi there, It is a kind of OT, but I couldn't find any useful information, and I already wasted half a day on this problem, so I thought someone here would be able to help me. It seems like my partition table is messed up, and I am not able to mount all my partitions. For example, mount refuse to mount /dev/hda12 on any directory: when I do a mount -r /dev/hda12 /mnt/tmp, mount tells me that hda12 is already mounted, or that /mnt/tmp is busy. The partition is not mounted for sure, and I tried several other tempory locations, without any success. The "funny" part is that a fsck.ext3 /dev/hda12 doesn't give me any error when I check the filesystem (which let me some hope about the possible recovery). Basically, I think the problem is only coming from a wrong partition table, but I don't know how to recover the good beginning/end of the partitions (the partitions used for the OS itself seem OK, my linux is works flawlessly, "only" my last data partitions are not accessible anymore). All my partitions are ext3. Thank, and my apologies for the OT, David

20 years, 2 months

[linux-audio-user] SimulAnalog Guitar suite VST plugin

by Pasi Karppinen

Hi! Anyone tried this promising free VST plugin under Wine? http://www.simulanalog.org/guitarsuite.htm "What is SimulAnalog Guitar suite? It is a suite of vst plugin with amplifiers and analog processors for guitarists, actually containing the simulation* of: - Boss DS-1 (Distortion stompbox) - Boss SD-1 (Super Overdrive stompbox) - Tube Screamer (Overdrive stompbox) - Oberheim PS-1 (Phaser stompbox) - Univox Univibe (Modulations stompbox) - Fender Twin 1969 (Guitar amplifier) - Marshall JCM900 Dual Reverb (Guitar amplifier)"

20 years, 2 months

[linux-audio-user] laptop concert

by Vedran Vucic

Hello all, Those who may be interested in laptop concerts may check www.laptopsounds.org So, you can check free software and participate in concert. Best regards, Vedran Vucic

20 years, 2 months

Re: [linux-audio-user] Speeding up Speech

by Dirk Jagdmann

Phillip Blevins wrote: > Ah, thanks for the suggestion, I had wondered about using a high > sample rate at high speed and then slowing it down. Any idea how > slowing the file down could be accomplished? The trick here is to just record and don't make any conversion. How will then just need to alter the WAV header to the correct sample rate (this can be done on good audio software). I'll explain this closer. We assume you have a tape running at 4x speed. Then you'll sample with 88.2KHz. If you would play back this recorded sample with 22KHz then it will be played 4x too slow. But has we have recorded it 4x too fast you'll have the correct speed. Every good audio/sample editor can adjust the sample rates *without* any conversion (so they just alter the values in the file headers). If you favourite sample editor insists on converting sample rates you should either get yourself a better program or just load a hex editor and change those 4 bytes for yourself (assuming .wav). > As for my track splitting question - I've found tracksplit > http://sourceforge.net/projects/tracksplit/ which allows me to > specify a minimum track lenth the and the amount of silence to split > on. This turns out to be perfect for voice wave data. [I posted this answer again to LAU, as I think other might be interested in my anser as well.] -- ---> doj / cubic ----> http://cubic.org/~doj -----> http://llg.cubic.org

20 years, 2 months

[linux-audio-user] SBlive Capture

by Mark Constable

This "trick" used to work a couple of years ago and now I want to do the same thing again but if I follow the example exactly I have no wav file output... http://alsa.opensrc.org/index.php?page=SBliveCapture I suspect the SBlive mixer settings are different these days and I'm hoping it's as simple as the right magic combo... would anyone know what that combo might be ? The other thing is now I have jack running but I'm not sure if that has anything to do with it... here's my aadebug output in case it's something silly with my own system and not my misunderstanding of the SBlive mixer controls... ALSA Audio Debug v0.0.8 - Mon Jan 31 19:49:46 EST 2005 http://alsa.opensrc.org/?aadebug Kernel ---------------------------------------------------- Linux shuttle 2.6.10 #2 SMP Sun Jan 23 10:22:10 EST 2005 x86_64 AMD Athlon(tm) 64 Processor 3000+ AuthenticAMD GNU/Linux Loaded Modules -------------------------------------------- snd_seq_midi 8128 0 snd_emu10k1_synth 8064 1 snd_emux_synth 37888 2 snd_emu10k1_synth snd_seq_virmidi 7680 1 snd_emux_synth snd_seq_midi_emul 9216 1 snd_emux_synth snd_emu10k1 89380 7 snd_emu10k1_synth snd_rawmidi 22944 3 snd_seq_midi,snd_seq_virmidi,snd_emu10k1 snd_ac97_codec 77984 1 snd_emu10k1 snd_util_mem 4864 2 snd_emux_synth,snd_emu10k1 snd_hwdep 9480 2 snd_emux_synth,snd_emu10k1 snd_seq_oss 33792 0 snd_seq_midi_event 8448 3 snd_seq_midi,snd_seq_virmidi,snd_seq_oss snd_seq 54720 15 snd_seq_midi,snd_emux_synth,snd_seq_virmidi,snd_seq_midi_emul,snd_seq_oss,snd_seq_midi_event snd_seq_device 8720 7 snd_seq_midi,snd_emu10k1_synth,snd_emux_synth,snd_emu10k1,snd_rawmidi,snd_seq_oss,snd_seq snd_pcm_oss 52900 0 snd_pcm 89864 5 snd_emu10k1,snd_ac97_codec,snd_pcm_oss snd_timer 22792 4 snd_seq,snd_pcm snd_page_alloc 9480 2 snd_emu10k1,snd_pcm snd_mixer_oss 18304 1 snd_pcm_oss snd 49992 25 snd_emux_synth,snd_seq_virmidi,snd_emu10k1,snd_rawmidi,snd_ac97_codec,snd_hwdep,snd_seq_oss,snd_seq,snd_seq_device,snd_pcm_oss,snd_pcm,snd_timer,snd_mixer_oss Proc Config ----------------------------------------------- CONFIG_SND=m CONFIG_SND_TIMER=m CONFIG_SND_PCM=m CONFIG_SND_HWDEP=m CONFIG_SND_RAWMIDI=m CONFIG_SND_SEQUENCER=m CONFIG_SND_OSSEMUL=y CONFIG_SND_MIXER_OSS=m CONFIG_SND_PCM_OSS=m CONFIG_SND_SEQUENCER_OSS=y CONFIG_SND_RTCTIMER=m CONFIG_SND_MPU401_UART=m CONFIG_SND_OPL3_LIB=m CONFIG_SND_VX_LIB=m CONFIG_SND_DUMMY=m CONFIG_SND_VIRMIDI=m CONFIG_SND_MTPAV=m CONFIG_SND_SERIAL_U16550=m CONFIG_SND_MPU401=m CONFIG_SND_AC97_CODEC=m CONFIG_SND_ALI5451=m CONFIG_SND_ATIIXP=m CONFIG_SND_ATIIXP_MODEM=m CONFIG_SND_AU8810=m CONFIG_SND_AU8820=m CONFIG_SND_AU8830=m CONFIG_SND_AZT3328=m CONFIG_SND_BT87X=m CONFIG_SND_CS46XX=m CONFIG_SND_CS46XX_NEW_DSP=y CONFIG_SND_CS4281=m CONFIG_SND_EMU10K1=m CONFIG_SND_KORG1212=m CONFIG_SND_MIXART=m CONFIG_SND_NM256=m CONFIG_SND_RME32=m CONFIG_SND_RME96=m CONFIG_SND_RME9652=m CONFIG_SND_HDSP=m CONFIG_SND_TRIDENT=m CONFIG_SND_YMFPCI=m CONFIG_SND_ALS4000=m CONFIG_SND_CMIPCI=m CONFIG_SND_ENS1370=m CONFIG_SND_ENS1371=m CONFIG_SND_ES1938=m CONFIG_SND_ES1968=m CONFIG_SND_MAESTRO3=m CONFIG_SND_FM801=m CONFIG_SND_FM801_TEA575X=m CONFIG_SND_ICE1712=m CONFIG_SND_ICE1724=m CONFIG_SND_INTEL8X0=m CONFIG_SND_INTEL8X0M=m CONFIG_SND_SONICVIBES=m CONFIG_SND_VIA82XX=m CONFIG_SND_VX222=m CONFIG_SND_USB_AUDIO=m CONFIG_SND_USB_USX2Y=m Modprobe Conf --------------------------------------------- alias char-major-14-* soundcore alias char-major-116-* snd alias snd-card-0 snd-emu10k1 alias sound-slot-0 snd-emu10k1 alias sound-service-0-0 snd-mixer-oss alias sound-service-0-1 snd-seq-oss alias sound-service-0-3 snd-pcm-oss alias sound-service-0-8 snd-seq-oss alias sound-service-0-12 snd-pcm-oss alias /dev/mixer snd-mixer-oss alias /dev/dsp snd-pcm-oss alias /dev/midi snd-seq-oss options snd cards_limit=1 Proc Asound ----------------------------------------------- Advanced Linux Sound Architecture Driver Version 1.0.6 (Sun Aug 15 07:17:53 2004 UTC). Compiled on Jan 23 2005 for kernel 2.6.10 (SMP). 0 [Live ]: EMU10K1 - Sound Blaster Live! Sound Blaster Live! (rev.6) at 0xc000, irq 17 1: : sequencer 0: [0- 0]: ctl 4: [0- 0]: hardware dependent 8: [0- 0]: raw midi 18: [0- 2]: digital audio playback 26: [0- 2]: digital audio capture 25: [0- 1]: digital audio capture 16: [0- 0]: digital audio playback 24: [0- 0]: digital audio capture 6: [0- 2]: hardware dependent 9: [0- 1]: raw midi 10: [0- 2]: raw midi 33: : timer 00-00: EMU10K1 (FX8010) 00-02: Emux WaveTable 00-00: emu10k1 : EMU10K1 : playback 32 : capture 1 00-01: emu10k1 mic : EMU10K1 MIC : capture 1 00-02: emu10k1 efx : EMU10K1 EFX : playback 8 : capture 1 Dev Snd --------------------------------------------------- controlC0 hwC0D2 midiC0D1 pcmC0D0c pcmC0D1c pcmC0D2p timer hwC0D0 midiC0D0 midiC0D2 pcmC0D0p pcmC0D2c seq CPU ------------------------------------------------------- model name : AMD Athlon(tm) 64 Processor 3000+ cpu MHz : 1994.868 RAM ------------------------------------------------------- MemTotal: 1025676 kB SwapTotal: 489940 kB Hardware -------------------------------------------------- 0000:00:00.0 Host bridge: nVidia Corporation nForce3 Host Bridge (rev a4) 0000:00:18.0 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] HyperTransport Technology Configuration 0000:00:18.1 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] Address Map 0000:00:18.2 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] DRAM Controller 0000:00:18.3 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] Miscellaneous Control 0000:01:05.0 Multimedia audio controller: Creative Labs SB Live! EMU10k1 (rev 06) --markc

20 years, 2 months

[linux-audio-user] Re: [Jackit-devel] Re: [linux-audio-dev] [ANNOUNCE] jack_convolve-0.0.1

by Steve Harris

On Mon, Jan 31, 2005 at 10:15:56 +0100, Alfons Adriaensen wrote: > On Sun, Jan 30, 2005 at 06:55:45PM -0600, Jack O'Quin wrote: > > > > In the case of JAMin it was dictated by our desire to use FFTW rather > > than writing our own FFT and trying to figure out how to divide the > > computation into relatively equal parts. > > For Jamin, you have two channels, each one needs a Fwd and an Inv FFT, > so that would already be 4 approx. equal parts (assuming the FFT dominates > the work to be done). Actually there is on fwd, and 3 inv per channel, each xover band is a seperate inv FFT. - Steve

20 years, 2 months

RE: [linux-audio-dev] Re: [Jackit-devel] Re: [linux-audio-user][ANNOUNCE] jack_convolve-0.0.1

by Reuben Martin

>>jack_convolve uses complex data format, not half-complex >>like brutefir. so i suppose i can't simply rip his code ;) >I'd recommend the half-complex format, it makes hte FFT much >faster and reduces the ammount of memory you have to touch. >The only pain is that the format is a bit wierd to work with. > >- Steve Forgive my ignorance, but is there any significant difference in the sound produced by this complex convolve and BruteFIR's half-complex convolve? (I'm speaking in a theoretical sense rather than comparing the actual implementations.) -Reuben

20 years, 2 months

[linux-audio-user] Re: [Jackit-devel] Re: [linux-audio-dev] [ANNOUNCE] jack_convolve-0.0.1

by Florian Schmidt

oops, sent to fons only.. here's the mail to the lists, too. sorry for duplicate, fons: On Sat, 29 Jan 2005 16:16:04 +0100 Fons Adriaensen <fons.adriaensen(a)skynet.be> wrote: > On Sat, Jan 29, 2005 at 03:43:46PM +0100, Florian Schmidt wrote: > > > Now i grok it. > > > > As a stereo impulse file is a response to a mono impulse placed > > somewhere in the room, one needs two stereo response files with the > > impulses created at different points in the room. > > Yep, that's it. Even for a stereo output, you could have more than > two inputs, each one corresponding to one place in the room. ack! > > > I will adapt the convolver to be able to load more than one response > > file and provide an input for each. And as many outputs as the response > > files have channels (for two stereo response files, that's two channels > > 'cause they are stereo).. > > That would do it. And then of course the input transform is done only > once, and the additions can be done in the F-domain, so there's also > only one output transform for each output. ack! The real cost is in the complex multiplications though. So if anyone knows how to speed that up with a nifty function using assembler, prefereably following this interface: // computes n complex multiplications and _add_s the results to the out buffer void complex_multiply(unsigned int n, fftwf_complex *in1, fftwf_complex *in2, fftwf_complex *out); i would be oh so greatful. So the TODO list as grown to: - fix error checking - add multiple response file support as explained above - add libsamplerate supprt for converting response files to jack samplerate - optimize the complex multiplication flo -- Palimm Palimm! http://affenbande.org/~tapas/

20 years, 2 months

Jump to page:

2025

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 January 2005