Linux-audio-dev

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3 participants
7520 discussions

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, 2 months

[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, 2 months

[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, 2 months

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, 2 months

[linux-audio-dev] Old tracks finaly recorded and mastered

by Thorsten Wilms

Hi! Finaly got around to record a number of old tracks that I created using nothing but a GEM S3 keyboard in 1997/98 (well, and my brother's PC for selecting samples and putting them on a floppy for importing). Previoulsy they only existed on floppy and tape and one floppy is corrupted already ... Still in the process of loading them up to archive.org, but some are there by now: In order of my own rating, best at the top: http://www.archive.org/download/s3_07_the_darkness_surrounding_me/thorwil_s… http://www.archive.org/download/s3_06_multibeatmode/thorwil_s3_06_multibeat… http://www.archive.org/download/s3_05_leaving_it_all_behind/thorwil_s3_05_l… http://www.archive.org/download/s3_04_red_sun/thorwil_s3_04_red_sun.ogg http://www.archive.org/download/s3_02_rave_scales/thorwil_s3_02_rave_scales… http://www.archive.org/download/s3_03_blue_beat/thorwil_s3_03_blue_beat.ogg http://www.archive.org/download/s3_08_where_do_i_want_to_go_today/thorwil_s… The detail pages can be reached from here: http://www.archive.org/search.php?query=creator%3A%22Thorsten%20Wilms%22 I wouldn't mind people writing reviews there ;) All recorded using the focused Timemachine. Cutting and normalizing with the most enjoyable Sweep. Mastering with the amazing Jamin. Need to practice that some more :). I wasn't aware of how much details can be brought to light using compression. The only problem was having to convert the 32bit float WAVs to 16 bit for lame, as it doesn't recognize that format. I used sndfile-convert for that. Too bad it doesn't take a flag for the output format, but requires filenames for that. But thanks to the nice folks at #lad, it didn't take long to arrive at a bash oneliner for converting a whole directoy. --- Thorsten Wilms

20 years, 2 months

[linux-audio-dev] Re: snd-usb-usx2y on amd64? - Success (more or less)

by Arnold Krille

Hi all, just a note: A got it more or less working. This morning I installed the vanilla-2.6.14-rc4 and I could start jackd in realtime with -n6 -p256 and run ardour and qsampler/linuxsampler in parallel without an xrun during normal operation. It just crashed jackd when I used -n6 -p128 but even after that it didn't ruin the sounddriver so I could start jackd again. That really takes a big stone from my heart as I have to do some serious music with the laptop this weekend and I feared I had to mess around with my old one... So thanks for your patience, Arnold -- visit http://dillenburg.dyndns.org/~arnold/ --- Wenn man mit Raubkopien Bands wie Brosis oder Britney Spears wirklich verhindern könnte, würde ich mir noch heute einen Stapel Brenner und einen Sack Rohlinge kaufen.

20 years, 2 months

[linux-audio-dev] Fixed vs. floating point

by cpolymeris＠gmx.net

I am in the early planning stage of an audio processing application and I have come to the point of making the choice between floating point or fixed point (signed 2's complement) processing. What do you think is better, and why? Why does jack use floating point? Why does AES use fixed point in most of their standards? I understand the technical difference between the formats, but... I can't come to a conclussion. The adavantages of floating point I see are: - easier processing, specially amplification and mixing (multiplying by fractions) - more sse support, more packed array arithmetics - relationship to voltage and SPL easier to understand. (1.0f vs. 2^31-1 max. values) And its diadvantages compared to fixed point would be: - generally slower, specially things like addition - 30 bits of resolution for 32 bits of data in IEEE single-precision (exponents larger than 0 not used for audio, neither are denormalized numbers and redundant zeroes, infinities, etc), compared to the full 32 bits of resolution of fixed point. - non-linear saw-tooth-like precission (precission gets higher as mantissa gets larger and then falls abruptly at the point the exponent is increased) I am sure I fail to see the more important points and would be thankful for any comments. If it has been discussed on the list earlier, i am sorry to post it again. I couldn't find it. Greetigs, Dimitri -- NEU: Telefon-Flatrate f�rs dt. Festnetz! GMX Phone_Flat: 9,99 Euro/Mon.* F�r DSL-Nutzer. Ohne Providerwechsel! http://www.gmx.net/de/go/telefonie

20 years, 2 months

[linux-audio-dev] Re: [ardour-dev] Proposed editing mode, for users with heavy editing needs

by Mike Fisher

Like said before good ideas. One thing that I agree with is to have the playhead snap to the beginning of a region when it is clicked on. This really would speed up the editing process. Maybe something like this could be toggled on or off in the options window. Another thing is in range mode. I'd like to have it so when you select a range that it would play that range just by hitting spacebar or play from the transport. As this would elimiate two mouse clicks. Seems small but it really. Maybe have this in the options window as well. Also in range mode. Move the play head with a single click would save alot of editing time. I know that if you hit p this will happen whereever the mouse is, but just clicking would be alot quicker. I'm not complaining about the existing operations. Ardour is awesome. I just like to be able to make quick edits. And eliminating clicks and keystrokes may not seem like much. But when you make possible hundreds of edits a day, it makes a huge difference in time. --- AES_24_96 <pipelineaudio(a)hotmail.com> wrote: > edit mode > > I propose an edit mode or maybe its a mouse mode > that would > allow editing tasks to be performed with MUCH higher > speed > and quite a few less keystrokes. A similar scheme > has been used > before to create an app with exponentially less > clicks and > keystrokes to perform many editing tasks than any > other. > > In this mode, playback cursor and edit cursor would > be > one and the same, a VERY skinny line that drops, and > stays > wherever you left click the mouse in the edit > window, either > on the ruler or on a region itself. > > Windows type conventions would apply to the > selection process. > Click a region, then shift click another region, and > those two > regions plus any region in between would be > selected. click a > region and control click another region for > selecting non > contiguous regions, and keep control clicking till > you have > selected all the regions you want. Click a fader and > shift click > or control click another fader to move multiple > faders at once. > > Simple stuff, some of which Ardour already has, m to > drop a > marker, s to split, r after making a range selection > to define a > storable range. "g" groups all selected regions > together, > whether on the same track or not, u ungroups > selected regions. > > Let me flesh some of these out a bit. For grouping, > all grouped > tracks would be moved as one of them is, either in > time or > between tracks. As groups overlap each other they > would be > auto-crossfaded where relevant, according to the > already > excellent Ardour autocrossfade options. Very > importantly, > there would be a "overide group" button in the > Ardour menu > bar, to perform actions on regions without bothering > other > regions grouped with them. Turning the button off > would > restore the group action. > > Splits can be made in several ways. Clicking on a > region and > hitting s will split right where the cursor sits. > Clicking in a > trackspace without a region in it, or along the > ruler then hitting > s will split all regions under the cursor. Dragging > a range > along the ruler then hitting s will put a split at > the very > beginning and end of the selected range. Selected > regions, > selected by click shift click and/or click control > click as > described above would also be split by hitting the s > key. > > Region editing would be similar to the way ardour is > now. > Resize by clicking and dragging on edges (would > group with > groups and selections). Control-click and drag on > edges will > time stretch or time compress, also groupable with > groups and > selections. Click dragging on the top edges of > either side will > control the fade in or fade out handle, same as > now. Click > dragging on the very top middle area of a region > would control > a per region volume trim. Right clicking a region > will bring up > a menu including things like reverse region, and > open copy of > region in audio editor. > > Mouse wheel editing would be somewhat different. > With no > modifier, scrolling the wheel will zoom time in and > out, > keeping the zooming centered around the cursor. In > the event > that zooming goes too far to keep centered, it will > go back to > the cursor center upon zooming back in. > > Snapping: Right clicking the "snap to" button brings > up a speed > menu where you can select whether to snap to ruler > marks > ( as defined in the ruler speed menu), time, SMPTE, > measures, half notes, quarter notes, 8th notes, 16th > notes, > 32nd notes, 64th notes and 128th notes). Holding > shift while > dragging a region overrides snapping. The "show > grid" next to > the snapping button will show a grid *VISIBLE > THROUGH > REGIONS* with its points set according to the "snap > to" > speed menu. > > Time display: Somewhere on ardour will be a time > details > screen. Top bar will show the timestamp of the > beginning of a > range selection. In the event of no range selection, > this will > show the timestamp at the cursor. The middle bar > will show > the timestamp at the end of the range selection. The > bottom > bar will show the duration of the range selection. > The main > time display as Ardour has now can be right clicked. > In the > speed menu that pops up, you can select between > SMPTE, > MTC, time(hour, minutes, seconds, miliseconds), > samples, > and musical time( measures, quarter notes, 8th > notes, etc...). > Separately the ruler can also be right clicked to > display the > same options. > > More to come, but I think with these basics ardour > would be > more than workable for those of Linux Noobs us who > have > been abandoned by our parent software company. > > I know it sounds like a lot, but I think this would > certainly > result in many fewer commands to accomplish the same > actions we have now, and actually be simpler to work > with > and use. > > _______________________________________________ > ardour-dev mailing list > ardour-dev(a)lists.ardour.org > http://lists.ardour.org/listinfo.cgi/ardour-dev-ardour.org > __________________________________ Yahoo! Music Unlimited Access over 1 million songs. Try it free. http://music.yahoo.com/unlimited/

20 years, 2 months

[linux-audio-dev] Documentation of the PADsynth synthesis algorithm

by Paul

Hi. I am Paul, the author of ZynAddSubFX software synthesizer. I wrote a complete documentation of how the PADsynth synthesis algorithm works. It includes the full description of the algorithm: overview, text description, diagrams, pseudocode; example C/C++ implementations and a ready-to-use C++ class. Also, I included some audio examples into the page. Please tell me your opinion about it. I tried to make it to be easy understood and ready to be implemented and/or used into your software synthesizers. Everything (the algorithm, the examples, the implementations, diagrams,etc) from this page are released under Public Domain. You are welcome to include and use this into your programs and I invite the companies/developers who write software synthesizers to use it. Also, I guess that the sound banks producers will be very interesed about this ;) I don't require anything for using it ;-) ; I just want this to be spread, to be used to make beautiful sounds. It's page is at: http://zynaddsubfx.sourceforge.net/doc/PADsynth/PADsynth.htm Enjoy! Paul __________________________________ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com

20 years, 2 months

Re: [linux-audio-dev] Documentation of the PADsynth synthesis algorithm

by Kjetil S. Matheussen

Paul: > Hi. > I am Paul, the author of ZynAddSubFX software > synthesizer. > I wrote a complete documentation of how the PADsynth > synthesis algorithm works. It includes the full > description of the algorithm: overview, text > description, diagrams, pseudocode; example C/C++ > implementations and a ready-to-use C++ class. > > Also, I included some audio examples into the page. > Please tell me your opinion about it. Wow, this is great work! Especially this one is so impressive: http://zynaddsubfx.sourceforge.net/doc/PADsynth/demos/0km.ogg --

20 years, 2 months

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Linux-audio-dev