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7535 discussions

[LAD] [LAA] [ANN] guitarix release guitarix2-0.18.0

by hermann

guitarix/gx_head, a guitar mono tube amplifier simulation for jack a new release (0.18.0) is available. please refer to our project page for more information: http://guitarix.sourceforge.net/ new features in short: * add tube model 12AT7 * fix runtime issues when build with g++ > 4.5 * add presence level controller * add bass booster level controller * switch to function pointer based engine * fix issues with GtkBuilder > 2.14 and Glade files * fix correct use of included zita-resampler source * add factory settings by autoandimat download site: http://sourceforge.net/projects/guitarix/ have fun Many thanks to Fons Adriaensen for lead me to find and fix a bogus bug related to correct object bindings in C++ kudos to Fons _________________________________________________________________________ As well, a new release (1.4) of gxtuner is available, gxtuner comes now with jack_session support. Read the README for details. http://sourceforge.net/projects/guitarix/files/gxtuner/ regards guitarix development team

14 years, 6 months

[LAD] BOSS BR-80 USB Audio

by Kazutomo Yoshii

Hi All, [ Sorry about reposting. Someone suggested LAD, so I reposted this to both LAD and LAU. ] I bought BOSS BR-80 recently, which can be used as USB audio interface. Unfortunately, Fedora15(2.6.38.8) didn't recognize it. Maybe someone already made BR-80 work but my google search didn't show any hint, so I decided to hack. I added an entry to quirks-table.h for BR-80 (see below) and recompiled the snd-usb-audio module and re-installed the module. I also added a dmex and softvol config to .asoundrc(see also below). It now works as a playback device, which satisfies my current requirements. The capture interface and MIDI seems not working. The lsusb output contains a couple of 'UNRECOGNIZED' lines. lsusb output,/proc/asound/, etc are also below. Although I might not use these interfaces, I'm just curious to hack. Can anyone suggest where I should look at? Thanks, Kaz ======================================== --- sound/usb/quirks-table.h.orig 2011-08-06 21:17:32.686663306 -0500 +++ sound/usb/quirks-table.h 2011-07-26 19:05:21.790767824 -0500 @@ -1611,6 +1611,34 @@ } } }, +{ + USB_DEVICE(0x0582, 0x0130), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + /* .vendor_name = "BOSS", */ + /* .product_name = "MICRO BR-80", */ + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 1, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 2, + .type = QUIRK_MIDI_STANDARD_INTERFACE + }, + { + .ifnum = -1 + } + } + } +}, + + ======================================== $HOME/.asoundrc pcm.dmixer { type dmix ipc_key 333444 # uniq val slave { pcm { type hw; card BR80AUDIO; } period_time 0 period_size 1024 buffer_size 8192 rate 44100 } bindings { 0 0 1 1 } } #http://alsa.opensrc.org/How_to_use_softvol_to_control_the_master_volume pcm.softvol { type softvol slave { pcm "dmixer" } control { name "Master" card 0 # is this right? } } pcm.!default { type plug slave.pcm "softvol" } ======================================== $ find /proc/asound/BR80AUDIO/ -type f /proc/asound/BR80AUDIO/id /proc/asound/BR80AUDIO/usbbus /proc/asound/BR80AUDIO/usbid /proc/asound/BR80AUDIO/stream0 /proc/asound/BR80AUDIO/midi0 /proc/asound/BR80AUDIO/pcm0p/sub0/status /proc/asound/BR80AUDIO/pcm0p/sub0/sw_params /proc/asound/BR80AUDIO/pcm0p/sub0/hw_params /proc/asound/BR80AUDIO/pcm0p/sub0/info /proc/asound/BR80AUDIO/pcm0p/xrun_debug /proc/asound/BR80AUDIO/pcm0p/info ======================================== This message appears in kernel log [81902.426215] ALSA sound/usb/midi.c:1680: MIDIStreaming interface descriptor not found ======================================== lsusb output Bus 002 Device 008: ID 0582:0130 Roland Corp. Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 255 Vendor Specific Class bDeviceSubClass 0 bDeviceProtocol 255 bMaxPacketSize0 64 idVendor 0x0582 Roland Corp. idProduct 0x0130 bcdDevice 0.00 iManufacturer 1 iProduct 2 iSerial 0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 176 bNumInterfaces 4 bConfigurationValue 1 iConfiguration 0 bmAttributes 0xc0 Self Powered MaxPower 2mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 255 Vendor Specific Subclass bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 2 iInterface 0 ** UNRECOGNIZED: 06 24 f1 01 00 00 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 2 iInterface 0 ** UNRECOGNIZED: 07 24 01 01 00 01 00 ** UNRECOGNIZED: 0b 24 02 01 02 04 18 01 44 ac 00 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x0d EP 13 OUT bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0038 1x 56 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 1 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 1 iInterface 0 ** UNRECOGNIZED: 07 24 01 07 00 01 00 ** UNRECOGNIZED: 0b 24 02 01 02 04 18 01 44 ac 00 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x8e EP 14 IN bmAttributes 37 Transfer Type Isochronous Synch Type Asynchronous Usage Type Implicit feedback Data wMaxPacketSize 0x0038 1x 56 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 0 bNumEndpoints 2 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 3 bInterfaceProtocol 0 iInterface 0 ** UNRECOGNIZED: 06 24 f1 02 01 01 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x03 EP 3 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 1 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x84 EP 4 IN bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 1 bNumEndpoints 2 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 3 bInterfaceProtocol 0 iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x03 EP 3 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 1 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x85 EP 5 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 1

14 years, 6 months

[LAD] LAD] RT-Safe UI/Engine Decoupling using Functional

by Jeff McClintock

I'm just in the process of porting my plugin to a platform where the GUI runs on Windows, but the Audio processing runs on an real-time optimized Linux box. With the proliferation of iPads etc, I see this approach of mixing a cool portable GUI with a Linux 'powerhouse' audio processor become more common in studios. Surly passing functors (which are pointers?) to audio code running in a separate address space can not work? Jeff McClintock > From: Florian Paul Schmidt <mista.tapas(a)gmx.net> > Subject: [LAD] RT-Safe UI/Engine Decoupling using Functional > Programming and Reference Counted SmartPointers > To: linux-audio-dev(a)lists.linuxaudio.org > Message-ID: <4E3CA5A4.7060208(a)gmx.net> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Hi, > > during the process of writing a new small jack sampler which fits my > workflow I came up with this little scheme to solve the UI/engine > decoupling problem. For the purpose of spreading the idea or > alternatively getting answers about how it's broken and sucks I decided > to write a little article describing it.. > > http://178.63.2.231/~tapas/wordpress/?page_id=45 > > The (largely unfinished and unusable) sampler project is here: > > https://github.com/fps/jass > > Let me have it.. > > Regards, > Flo

14 years, 6 months

Re: [LAD] [LAU] Announcing Kn0ck0ut-LV2, spectral subtraction plugin

by Jeremy Salwen

Thank you, I apologize. That is now fixed. Jeremy On Sun, Aug 7, 2011 at 12:09 PM, Alvaro Paz da Silva <alanpasi(a)gmail.com>wrote: > Hi > File not found... :( > https://nodeload.github.com/jeremysalwen/kn0ck0ut-LV2/tarball/1.0 > > Alanpasi > > On Sun, Aug 7, 2011 at 6:58 AM, Jeremy Salwen <jeremysalwen(a)gmail.com>wrote: > >> Hello all, >> >> I am pleased to announce that the first release of Kn0ck0ut-LV2<https://github.com/jeremysalwen/kn0ck0ut-lv2>is ready. Kn0ck0ut-LV2 is a port of the very popular VST "Spectral >> Subtractor/Vocal Remover" plugin Kn0ck0ut<http://www.freewebs.com/st3pan0va/>, >> with added features and improvements. >> >> Kn0ck0ut-LV2 is primarily useful for either isolating or removing the >> "center channel" of stereo recordings. If you have center panned vocals, >> you can use this to get the classic effects of "karaoke" or "vocal >> extraction". However, if you turn the controls to the extremes and/or feed >> it two unrelated signals, you can also get a variety of unique sounds. >> >> On top of Kn0ck0ut's functionality, I offer these improvements in >> Kn0ck0ut-LV2: >> >> * Improved performance through use of FFTW and more efficent buffering code >> * Completely variable FFT size and overlap amount controls. >> >> >> >> >> * Experimental "Phase Compensation" option, which will perhaps preserve >> additional fidelity in certain cases. >> * Restored Low-cut filter which was removed in later releases. >> >> If this all sounds great, a direct download link for the sources is here<https://github.com/jeremysalwen/kn0ck0ut-LV2/tarball/1.0>. >> In order to compile, you need lv2core, fftw3, and lv2-c++-tools >> >> A special thanks to St3pan0va, the original author of kn0ck0ut. >> >> Comments/questions are welcome. >> >> Jeremy Salwen >> >> _______________________________________________ >> Linux-audio-user mailing list >> Linux-audio-user(a)lists.linuxaudio.org >> http://lists.linuxaudio.org/listinfo/linux-audio-user >> >> >

14 years, 6 months

[LAD] [LAU] Announcing Kn0ck0ut-LV2, spectral subtraction plugin

by Jeremy Salwen

Hello all, I am pleased to announce that the first release of Kn0ck0ut-LV2<https://github.com/jeremysalwen/kn0ck0ut-lv2>is ready. Kn0ck0ut-LV2 is a port of the very popular VST "Spectral Subtractor/Vocal Remover" plugin Kn0ck0ut<http://www.freewebs.com/st3pan0va/>, with added features and improvements. Kn0ck0ut-LV2 is primarily useful for either isolating or removing the "center channel" of stereo recordings. If you have center panned vocals, you can use this to get the classic effects of "karaoke" or "vocal extraction". However, if you turn the controls to the extremes and/or feed it two unrelated signals, you can also get a variety of unique sounds. On top of Kn0ck0ut's functionality, I offer these improvements in Kn0ck0ut-LV2: * Improved performance through use of FFTW and more efficent buffering code * Completely variable FFT size and overlap amount controls. * Experimental "Phase Compensation" option, which will perhaps preserve additional fidelity in certain cases. * Restored Low-cut filter which was removed in later releases. If this all sounds great, a direct download link for the sources is here<https://github.com/jeremysalwen/kn0ck0ut-LV2/tarball/1.0>. In order to compile, you need lv2core, fftw3, and lv2-c++-tools A special thanks to St3pan0va, the original author of kn0ck0ut. Comments/questions are welcome. Jeremy Salwen

14 years, 6 months

[LAD] prototype in header but declaration where?

by Renato

Hi, very noob C question. I do this: renato@acerarch /usr/include/alsa $ grep snd_seq_open * seq.h:int snd_seq_open(snd_seq_t **handle, const char *name, int streams, int mode); seq.h:int snd_seq_open_lconf(snd_seq_t **handle, const char *name, int streams, int mode, snd_config_t *lconf); so I see that the function snd_seq_open has its prototype declared in seq.h... but where is the actual function definition? cheers renato

14 years, 6 months

[LAD] [ANN] International Conference on Spatial Audio, Nov 10-13 2011, in Detmold, Germany

by Jörn Nettingsmeier

hi *! sorry for the slightly off-topic post, but since spatial audio has been a frequent topic lately, i think some people here might be interested. linux or FLOSS won't be exactly in the limelight, but yours truly will make sure there are at least 2-3 boxes with your favourite OS and audio tools humming along in various places. oh, and you might come early and watch a few high-end mixing consoles boot - the startup screen will bring tears to your eyes (as will the price tag, unfortunately :) unfortunately, there will have to be an admission fee, which we haven't decided on yet. but we're trying to keep it reasonable. don't shout at me when it turns out to be a bit more costly than LAC, though... jörn *.* ICSA 2011 - International Conference on Spatial Audio November 10 - 13, Hochschule für Musik, Detmold Organizers: Verband Deutscher Tonmeister (VDT), in cooperation with Deutsche Gesellschaft für Akustik e.V. (DEGA), and European Acoustics Association (EAA). Contact/Chair: Prof. Dr.-Ing. Malte Kob Erich-Thienhaus-Institut Neustadt 22, 52756 Detmold Mail: icsa2011attonmeister.de Phone: +49-(0)5231-975-644 Fax: +49-(0)5231-975-689 Summary: The International Conference on Spatial Audio 2011 takes place from November 10 to 13 at Detmold University of Music. This expert‘s summit will examine current systems for multichannel audio reproduction and complementing recording techniques, and discuss their respective strengths and weaknesses. Wavefield synthesis systems, a higher-order Ambisonics array, as well as 5.1/7.1 installations in diverse acoustic environments will be available for comparative listening tests during the conference. Structured plenary talks, paper and poster sessions will revisit fundamentals and present latest research. A series of workshops will be dedicated to practical implementations of spatial sound capture and playback methods, and their esthetic and psychoacoustical implications for music perception. Concerts that include music specially arranged for the conference will let you experience various spatial sound systems in "live" conditions. Call for papers and music: Your contributions are welcome, either as presentations, posters, or workshops. Submissions will undergo a review process, and accepted contributions will be published in the conference proceedings. The conference language is English. We are planning structured sessions on the following topics: * Multichannel stereo * Wave field synthesis * Higher-order Ambisonics / spherical acoustics * 3D systems * Binaural techniques An additional session will be dedicated to related miscellaneous contributions, such as hybrid systems and perception/evaluation of spatial music reproduction.

14 years, 6 months

[LAD] Help with ALSA plugin - adding real time parameter

by Bob DeCarlo

Hi, I am trying to get my own ALSA plug-in to work with some real time controls. ( Is this the right place to ask?) I am successful with the PCM part of it. What I mean is from: http://www.alsa-project.org/alsa-doc/alsa-lib/pcm_external_plugins.html External Plugin: Filter-Type Plugin static snd_pcm_sframes_t myplg_transfer(snd_pcm_extplug_t *ext, const snd_pcm_channel_area_t *dst_areas, snd_pcm_uframes_t dst_offset, const snd_pcm_channel_area_t *src_areas, snd_pcm_uframes_t src_offset, snd_pcm_uframes_t size) { // my PCM processing works // I want to add a control parameter that can be set in real time by an app } SND_PCM_PLUGIN_DEFINE_FUNC(myplg) { // This all works for PCM processing just like the examples ... err = snd_pcm_extplug_create(&mix->ext, name, root, sconf, stream, mode); if (err < 0) { free(mix); return err; } ... } SND_PCM_PLUGIN_SYMBOL(myplg); Now I want to add a simple real time adjustment, an integer value that can be sent by an application to adjust the sound (PCM samples) in real time. I tried some ways of doing it it but without success, I am not understanding the basics. I first looked at doing a ctl. I was able get separately (without PCM processing) get a control to work but that looks like ts only for hardware control? I can't connect it to my PCM processing. http://www.alsa-project.org/alsa-doc/alsa-lib/ctl_external_plugins.html I looked at LADSPA but I'm not sure where that is going to take me. I am now looking at http://www.alsa-project.org/alsa-doc/alsa-lib/pcm_external_plugins.html External Plugin: I/O Plugin But, I am confused because I also see pcm type callbacks. So, I do not have a specific coding question (yet) but I just need a general direction. ... -Should I use ctl_external_plugins and figure out how to use it with my PCM? -Should I use a LADSPA example? -Should I go wth External Plugin: I/O Plugin? -Something else? I hope I am clear enough about my question and thanks for any pointers you can provide. Bob

14 years, 6 months

[LAD] excellent overview article on realtime audio programming

by Paul Davis

Ross Bencina is the author of AudioMulch and has been extremely involved in PortAudio, ReacTable, and other projects. His new article on realtime audio programming is a MUST read for anyone new to the area, and worth reading as a reminder even for experienced developers. http://www.rossbencina.com/code/real-time-audio-programming-101-time-waits-… I'm off to count how many violations Ardour contains ... --p

14 years, 6 months

[LAD] New Java binding for JACK

by Neil C Smith

Hi All, Just a quick post to let you know about a new JACK binding for Java, called JNAJack, at http://code.google.com/p/java-audio-utils/ JNAJack is a minimal object-oriented wrapper to the JACK Audio Connection Kit API. It uses Java Native Access (JNA) rather than custom JNI to interface with the underlying Jack API, simplifying development and deployment - no compilation required and it (*should*) work cross platform. Use of JNA means that performance is not quite on a par with the custom JNI code in JJack, but it is still fine for low latency usage, and some further performance optimisations are in the wings. Unlike JJack the aim of this project is to support full and typesafe OOP access to the Jack API from Java, and nothing else. Most important aspects of the audio API are included. MIDI and transport support will be implemented in the future. Well, I say this is new, it was mostly written quite a while back as part of my Praxis InterMedia System project http://code.google.com/p/praxis/ (a Java based cross-media patcher environment). I'm finally getting around to releasing some of this stuff separately as well. Best wishes, Neil -- Neil C Smith Artist : Technologist : Adviser http://neilcsmith.net

14 years, 6 months

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Linux-audio-dev