Linux-audio-dev

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7516 discussions

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

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

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

[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

[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

[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

Re: [linux-audio-dev] Edirol FA-101

by Dmitry S. Baikov

> Are you using a customized jackd? What version? What command line? Do > you have any evidence that anyone has ever made this work? > Opps, sorry for skipping obviously needed details. Was really upset. I tried freebob + jackd from freebob.sf.net. libavc from svn, libiec61883 1.0, libraw1394 1.2 cmdline: jackd -d iec61883 -o osc.udp://localhost:31000 FreeBoB wiki's list of working setups contains FA-101 + gentoo (my distro). When run in the first time, jackd starts, but there's no sound, and seems processing callbacks aren't called (no interrupts?). Dmitry.

20 years

[linux-audio-dev] Transport Regions

by Thorsten Wilms

Hi! I have been thinking about combining sequencing, (live) looping and sampling. I call the concept I arrived at Transport Regions for now (might call it Areas, Scopes, Frames ... a native speaker's take on this?). A Transport Regions groups n tracks, defining a common playback posistion and transport state (playing, paused, reverse ...). Loops can be defined with Markers. A 'classic' sequencer/daw project would use only 1 Region, but having several could allow sooperlooper like action, with the advantage of simple extension, moving on from jamming to production. Transport commands to specific Regions could be recorded and played back themselves. Instrument type samples could be treated the same, with the known markers, just for rather short loops, provided transport actions could be mapped to midi/notes. Multisampling would require means to map (midi) parameters to track level changes and/or soloing/muting. Patterns could actualy be Transport Regions triggered from a track, from which they would need to 'inherit' tempo for normal operation. Well ... just food for thought! --- Thorsten Wilms

20 years

[linux-audio-dev] Edirol FA-101

by Dmitry Baikov

Hello! Recently got FA-101. But can't get it to work. Jackd log is below. Thank you in advance. Regards, Dmitry. ---- JACK compiled with System V SHM support. loading driver .. IEC61883: Using FreeBobCtl lib version 0.0.1 (Build Oct 20 2005 - 01:33:14) Reading from OSC URL: osc.udp://localhost:31000 requesting channel info from osc.udp://localhost:31000 adding generic handler... adding /reponse handler... parsing address... sending request... waiting for response... ok freeing... done... ConnectionInfo contains 1 connection_specs IEC61883: Adding 1 capture connection(s)... IEC61883: capture connection 0: 0 streams, dim 0 from (0,11,11) IEC61883: adding stream 0: (1,1) 0x06 0x02 -> 0 (MicIn1 l) IEC61883: adding stream 1: (6,2) 0x06 0x02 -> 0 (MicIn1 right) IEC61883: adding stream 2: (2,1) 0x06 0x03 -> 0 (LineIn 3&4 l) IEC61883: adding stream 3: (7,2) 0x06 0x03 -> 0 (LineIn 3&4 right) IEC61883: adding stream 4: (3,1) 0x06 0x03 -> 0 (LineIn 5&6 l) IEC61883: adding stream 5: (8,2) 0x06 0x03 -> 0 (LineIn 5&6 right) IEC61883: adding stream 6: (4,1) 0x06 0x03 -> 0 (LineIn 7&8 l) IEC61883: adding stream 7: (9,2) 0x06 0x03 -> 0 (LineIn 7&8 right) IEC61883: adding stream 8: (0,1) 0x06 0x04 -> 0 (SpdifIn left) IEC61883: adding stream 9: (5,2) 0x06 0x04 -> 0 (SpdifIn righ) IEC61883: adding stream 10: (10,1) 0x0D 0x0A -> 0 (MidiPort) requesting channel info from osc.udp://localhost:31000 adding generic handler... adding /reponse handler... parsing address... sending request... waiting for response... ok freeing... done... ConnectionInfo contains 1 connection_specs IEC61883: Adding 1 playback connection(s)... IEC61883: playback connection 0: 0 streams, dim 0 to (0,11,11) IEC61883: adding stream 0: (1,1) 0x06 0x03 -> 0 (LineOut 1&2 left) IEC61883: adding stream 1: (6,2) 0x06 0x03 -> 0 (LineOut 1&2 righ) IEC61883: adding stream 2: (2,1) 0x06 0x03 -> 0 (LineOut 3&4 left) IEC61883: adding stream 3: (7,2) 0x06 0x03 -> 0 (LineOut 3&4 righ) IEC61883: adding stream 4: (3,1) 0x06 0x03 -> 0 (LineOut 5&6 left) IEC61883: adding stream 5: (8,2) 0x06 0x03 -> 0 (LineOut 5&6 righ) IEC61883: adding stream 6: (4,1) 0x06 0x03 -> 0 (LineOut 7&8 left) IEC61883: adding stream 7: (9,2) 0x06 0x03 -> 0 (LineOut 7&8 righ) IEC61883: adding stream 8: (0,1) 0x06 0x04 -> 0 (SpdifOut lef) IEC61883: adding stream 9: (5,2) 0x06 0x04 -> 0 (SpdifOut rig) IEC61883: adding stream 10: (10,1) 0x0D 0x0A -> 0 (MidiPort) IEC61883D: Creating driver (period_size=256, ringbuffer_size=8192) Creating IEC61883 client... 256/8192/48000 poll timeout = 5 ms Creating 11 buffers of 8192 quadlets... IEC61883C: Creating playback connection from node 65472, plug 0 prebuffers=0, buffers=300, irq_interval=100 Creating 11 buffers of 8192 quadlets... IEC61883 Client created... Initializing OSC debug & statistics server... port 31001 OK Starting OSC debug & statistics server... OK IEC61883CM: registered jack port cap_0_0_0_MicIn1 l IEC61883CM: registered jack port cap_0_0_0_MicIn1 right IEC61883CM: registered jack port cap_0_0_0_LineIn 3&4 l IEC61883CM: registered jack port cap_0_0_0_LineIn 3&4 right IEC61883CM: registered jack port cap_0_0_0_LineIn 5&6 l IEC61883CM: registered jack port cap_0_0_0_LineIn 5&6 right IEC61883CM: registered jack port cap_0_0_0_LineIn 7&8 l IEC61883CM: registered jack port cap_0_0_0_LineIn 7&8 right IEC61883CM: registered jack port cap_0_0_0_SpdifIn left IEC61883CM: registered jack port cap_0_0_0_SpdifIn righ IEC61883CM: registered midi port MidiIn_0_0_0_MidiPort_1 as 128:0 IEC61883CM: registered jack port pbk_0_0_0_LineOut 1&2 left IEC61883CM: registered jack port pbk_0_0_0_LineOut 1&2 righ IEC61883CM: registered jack port pbk_0_0_0_LineOut 3&4 left IEC61883CM: registered jack port pbk_0_0_0_LineOut 3&4 righ IEC61883CM: registered jack port pbk_0_0_0_LineOut 5&6 left IEC61883CM: registered jack port pbk_0_0_0_LineOut 5&6 righ IEC61883CM: registered jack port pbk_0_0_0_LineOut 7&8 left IEC61883CM: registered jack port pbk_0_0_0_LineOut 7&8 righ IEC61883CM: registered jack port pbk_0_0_0_SpdifOut lef IEC61883CM: registered jack port pbk_0_0_0_SpdifOut rig IEC61883CM: registered midi port MidiOut_0_0_0_MidiPort_1 as 128:1 Client start... creating capture connections... libiec61883 warning: iec61883_cmp_create_p2p_output: Failed to set the oPCR[0] plug f or node 0. Init ISO master receive handler on channel -1... (BUFFER=300,PACKET_MAX=2048 ,IRQ=100)... Start ISO master receive... IEC61883C: couldn't start receiving: Invalid argument DRIVER NT: could not start driver cannot start driver

20 years

Re: [linux-audio-user] Re: [linux-audio-dev] Edirol FA-101

by Lee Revell

On Fri, 2005-10-21 at 14:48 -0700, Mark Knecht wrote: > > I think this would be a better question for the Freebob list, and cc: > > the jackit-devel list, as you're using a version of JACK that the > > Freebob people have customized. I've never heard anyone on LAD or LAU > > report that this works. > > > > First and foremost, we need to get the iec61883 driver into JACK CVS, so > > that Paul Davis and the other JACK experts can help you. > > > > Lee > > In case some folks don't know this stuff iec61883 is part of the 1394 > stack. Why would it go into Jack CVS? Sorry, I mean "the iec61883 backend". JACK used to call these "drivers" but as you can see it's confusing. JACK does not need to include the iec61883 stack but it does need to know how to talk to it, just like with ALSA, OSS, etc. Look at his command line: jackd -d iec61883 -o osc.udp://localhost:31000 If I run this I get: rlrevell@mindpipe:~/kernel-source/linux-2.6.13$ jackd -d iec61883 jackd: unknown driver 'iec61883' So he must be using a third party patch to jackd from the Freebob people that implements the 'iec61883' backend. Lee

20 years

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