Linux-audio-dev

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3 participants
7520 discussions

[linux-audio-dev] Latency and feedback problems: soundcard with live microphone pass-thru, optimal solution ?

by Benno Senoner

Hi all, I would like to route a microphone through a sound card and back to powerful amplified speakers. As we know in analog PA gear you have the microphone feedback problem (usually it comes in form of high pitched whistle sounds). But if I route a mic from into the soundcard and out to the speakers, there will be a small delay due to the audio card buffers. Even if it's only a few msecs it makes the problem much worse than in the case of analogue gear because the feedback sound will come in chunks that's one audio card buffer at time. For example if I use 64 samples per buffer which gives me acceptable latency for a live singer, the feedback noise could possibly generate a much lower pitched signal/interference (I assume something like 44100/64 Hz) which is I think not easy to filter out compared to the high pitched feedback (is in the latter case sufficient to cut some high frequencies using an EQ ?). How can one solve the feedback problem in case of mic to speaker delays of let's say 5-10msecs ? Is an echo canceller algorithm needed ? If yes does this compromise the quality of the input signal (the singer). I think this is a very interesting topic and it would be cool if knowledgable people could come up with topics and ideas. (for example people that are good at DSP, room correction etc, like Fons A. etc). PS: I know that cards like DELTA 1010 and others (RME) can do zero latency monitoring (hardware pass-thru) but I'd prefer a software based routing since you can apply effects and stuff before forwarding the output. thanks for infos and toughts Benno http://www.linuxsampler.org

20 years, 1 month

[linux-audio-dev] [ANN] WhySynth DSSI softsynth

by Sean Bolton

Introducing WhySynth, a DSSI softsynth plugin. WhySynth, as in 'Y'-synth, the super-sized, frankensteinized, evolved and mutated, still rather dorky younger sibling of Xsynth-DSSI. WhySynth, as in (I sometimes ask), "_why_ am I working on another softsynth instead of on paying gigs?" (Following my bliss? Addiction? One last shot at misspent youth?) WhySynth, as in a mostly-new design featuring: - 4 oscillators per voice, in your choice of 6 modes (minBLEP, wavecycle, asynchronous granular, FM, waveshaper, and noise), - 2 filters, also in multiple flavors, - flexible routing and mixdown to stereo output, - 3 (or is it 6?) LFOs (instrument-wide, per-voice, and multiphase), - 5 multi-mode envelope generators, - abundant modulation options, - and effects (well, Tim Goetze's Versatile plate reverb is all at the moment, unless you count the DC-blocker anti-effect). WhySynth is a work in progress. Actually, since the kid was born, progress has slowed to a near-utter standstill, but if I can't release often, I might as well release early. Get your tarball, boring screenshot, and html-ized README today at: http://home.jps.net/~musound/whysynth.html then get your butts back to making cool music -- however you define that. Cheers, -Sean

20 years, 1 month

[linux-audio-dev] Radio receiver.

by James Courtier-Dutton

Hi, I want to plug a ham radio receiver into the sound card of my PC, and use the CPU of the PC to tidy up the signal, to hopefully make it more readable. One form of communications is called Morse Code, where a single tone is switched on and then off to pass a signal over the radio. I know it is easy to do band pass filters in Linux so that only the tone gets through, but the other really useful thing would be a noise filter. I was hoping that someone had already written a open source noise filter that I could use. If so, can someone point me to it, because I can't seem to find it on google. The noise filter needs to happen in real time, so ideally a plugin for ardour of something like that. James

20 years, 1 month

[linux-audio-dev] ladspa problem on OSX 10.4 figured out

by Taybin Rutkin

On 10.4, Apple removed the deprecated dlopen() mechanism of using _init() and _fini(). Instead, function attributes should be used. These have been supported in gcc since at least 2.9x. So instead of: void _init() {} void _fini() {} you should use: __attribute__((constructor)) void init() {} __attribute__((destructor)) void fini() {} Thanks, Taybin

20 years, 1 month

RE: [linux-audio-dev] Re: applying RIAA curves in software

by Cornell III, Howard M

I stand by my assertion that the RIAA record curve attenuates the low frequencies and amplifies the high frequencies. The physical effects on the record are such that the attenuated low frequencies do not cause the cutting head trace to take up so much room on the master, and by the same reasoning, the normally low amplitudes at high frequencies are given more (physical) headroom. Of course the reverse filter would flatten the frequency response and the use of the record filter optimizes the use of the master "real estate". -----Original Message----- From: linux-audio-dev-bounces(a)music.columbia.edu [mailto:linux-audio-dev-bounces@music.columbia.edu] On Behalf Of fons adriaensen Sent: Tuesday, October 25, 2005 5:00 PM To: The Linux Audio Developers' Mailing List Subject: Re: [linux-audio-dev] Re: applying RIAA curves in software On Tue, Oct 25, 2005 at 04:10:12PM -0500, Richard Smith wrote: > > > > > The RIAA record curve reduces bass and increases treble, and the > > > reverse RIAA curve for playback does the opposite. > > > > Sorry, but this is plain wrong. The RIAA filter used when cutting a > > disk master will boost bass (below 50 Hz), and reduce high > > frequencies. This actually leads to a worse S/N ratio on playback. It looks like this: > > I'm confused then. > > This page: > > http://www.bonavolta.ch/hobby/en/audio/riaa.htm > > Has a spread sheet that runs the math on the equation presented. > Unless I'm just backwards for RIAA reproduction it yeilds roughly 20dB > of gain for 20Hz and -21dB for 21kHz. Which seems backward from what > you are saying. That curve, the same as (3) in my previous post, is often called 'the RIAA curve' but it isn't. It is the combination two things: * a 1/F (or -6dB/oct) filter that is required to compensate for the +6dB/oct response of a magnetic cartridge, * and the real RIAA playback curve, (2) in my previous post, which boosts high frequencies. In other words, the general downward slope of the filter you refer to has nothing to do with RIAA equalisation, it's there only because that filter is designed for use in a preamp for a magnetic cartridge. If you would have a flat frequency response from the cartridge, then the filtering required needs to boost high frequencies, as in (2). If you take (2), and turn it 45 degrees clockwise (that adds the -6dB/oct slope for the transducer), you get the curve you refer to. So the idea that the RIAA curve was introduced to improve the S/N ratio at high frequencies is just wrong, it does the opposite. -- FA

20 years, 1 month

RE: [linux-audio-dev] Re: applying RIAA curves in software

by Cornell III, Howard M

Maybe use a crystal cartridge that puts out approximately one volt instead of a magnetic cartridge that puts out tens of millivolts. That attacks the signal to noise problem from the signal side. The RIAA record curve reduces bass and increases treble, and the reverse RIAA curve for playback does the opposite. If most of the objectionable noise is high frequency and you only filter it once wouldn't you just have to do it when the noise was accentuated, before the reverse RIAA filter? -----Original Message----- From: linux-audio-dev-bounces(a)music.columbia.edu [mailto:linux-audio-dev-bounces@music.columbia.edu] On Behalf Of Richard Smith Sent: Tuesday, October 25, 2005 1:36 PM To: The Linux Audio Developers' Mailing List Subject: Re: [linux-audio-dev] Re: applying RIAA curves in software > Quite a lot of words. I´m impressed. And convinced ... > > ... of the fact, that these guys just want to sell their stuff. > You really think its just all hype? Sort of made sense to me. If you do click and pop removal prior to the un-RIAA. Then that will go that much further to reduce the noise since its all high-frequency. > If you have a decent phono preamp, use it and forget all about that > tracertek advertising hype. > I don't have a preamp at all. Thats part of what led me to that page. Recommendations? -- Richard A. Smith

20 years, 1 month

Re: [linux-audio-dev] applying RIAA curves in software

by Richard Audette

Not sure if it's scriptable, but I believe Audacity has this built in: Effects -> Equalization -> RIAA Richard At 09:38 AM 10/25/2005, you wrote: >I'm going to convert my fathers record collection over to CD. Doing >some google research. > >According to http://www.tracertek.com/newway.htm they claim the "new" >and best way to do LP to CD is to use a flat preamp, record at 24bit, >96kHz and then apply the RIAA curve in software after the fact. >Either before or after the DeNoise, De-Click, etc depending. I've >also seen a few other sites that say the same type things. > >tracertek sells doze software to do the whole ball of wax but I'd like >to use Linux. > >I haven't found any RIAA filters yet so I guess I'm looking at >writeing one. So does anyone have any information on where to find >the official RIAA curve to make a plugin from? > >They also recommend using a pink-noise record to calibrate your setup >and then adjust the curve so it matches your system. > >-- >Richard A. Smith

20 years, 1 month

[linux-audio-dev] Re: [linux-audio-user] Edirol FA-66 Working Setup

by Bradley A. Hare

ce wrote: >there was a talk at LAC 2005; AFAIR the latency was 50msecs this time. >I don't know if it is less these days. I believe I am at 10 msec. now - have not pushed it further yet. >are they already working on a true ALSA driver? I believe they are. There is a dev. road map on the sourceforge page. Regards, Brad Hare

20 years, 2 months

[linux-audio-dev] 22050hz mono volume fading

by Ross Levis

It's my first post here. I'm developing an audio player which has a fading up/down facility. This is working fine except for 22050 mono samples. I'm getting patchy loud noises (intermittent white noise?) while fading the audio data up and down. The original audio can also be heard between the noise. The basic equation I'm doing is (Excuse the Pascal syntax): Data := Data * CurrVol / 32767; Data is always 2 bytes of audio data (16 bit data). CurrVol goes from 0 to 32767 or vice versa over several bytes. If I remove this line, there is no noise generated, but obviously no volume change either. The problem only occurs when it's mono 22050 samples. Stereo 22050 samples are fine, and mono 44100 samples are fine. I don't understand why this is happening. Any ideas? Thanks, Ross.

20 years, 2 months

[linux-audio-dev] [ANN] Sineshaper 0.4.0

by Lars Luthman

The Sineshaper is a monophonic DSSI synth. This is the first release. Source tarball, screenshot and Vorbis demo are available here: http://ll-plugins.sf.net. The knob graphics are created by Thorsten Wilms and Peter Shorthose. The Sineshaper synth has two sine oscillators and two waveshapers. The sound from the two oscillators is mixed and passed through the waveshapers, first through the first waveshaper and then the second. You can control the tuning of both oscillators as well as their relative loudness, and the total amount of shaping and the fraction of that amount that each shaper applies. Both waveshapers use a sine function for shaping the sound, but for the second shaper you can shift the sine function (with maximal shift it becomes a cosine function) to produce a different sound. You can also add vibrato and tremolo, and change the ADSR envelope that controls the amplitude and shape amount (as well as setting the envelope sensitivity for both the amplifier and the shapers). There is also a "Drive" control that adds distorsion, and a feedback delay with controllable delay time and feedback amount. All control parameters can be changed using MIDI. The Sineshaper synth comes with some presets that you can play or use as starting points for your own synth settings. You can not change these "factory presets", but you can create and save your own presets. They are written to the file .sineshaperpresets in your home directory. If you make any nice presets I would really like to hear them. -- Lars Luthman PGP key: http://www.d.kth.se/~d00-llu/pgp_key.php Fingerprint: FCA7 C790 19B9 322D EB7A E1B3 4371 4650 04C7 7E2E

20 years, 2 months

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Linux-audio-dev