Linux-audio-dev

linux-audio-dev@lists.linuxaudio.org

4 participants
7502 discussions

by Aurélien Leblond

Hi everyone, In my lv2 plugin, I need to communicate information from the DSP to the UI, but I don't want to break the DSP/UI separation principle (no Instance or Data access). On top of that, I'm using LVTK. The type of data I'm trying to move is a Vector of Float (buffer type Sound in the ttl) - but even basic information on how to transfer a Float from the DSP to the UI would be valuable. What I have do so far in the DSP (run method) m_forge->set_buffer((uint8_t*)p(p_notify), sizeof(float)); //p_notify being the atom port m_forge->sequence_head(m_notify_frame, 0); ForgeFrame p_frame; Atom p_msg = m_forge->write_blank(p_frame, 1, urids.atom_Float); m_forge->property_head(urids.atom_value, 0); m_forge->write_float(1604); //test data i'm trying to move m_forge->pop(p_frame); In my UI, I just want to see if I receive this data In the port_event method, at the moment the only thing I'm doing is: std::cout << port << " - " << format << std::endl; The port is defined in the TTL as follows: a lv2:OutputPort, atom:AtomPort ; atom:bufferType atom:Float ; atom:supports <http://lv2plug.in/ns/ext/patch#Message> ; lv2:index 1 ; lv2:symbol "notify" ; lv2:name "Notify" ; At the moment, the code compiles and runs without crashing (yeah!), but nothing happens (boo!). Would you guys have any ideas? Thanks in advance, Aurélien

11 years, 8 months

linux-port of OpenMusic

by anders.vinjar＠bek.no

Hi all. Ive been working on a linux-port of IRCAMs OpenMusic lately, and think its approaching a useful state now. I would very much appreciate it if some of the brave ones out there did some testing and sent me feedback: http://forge.ircam.fr/p/OM6/downloads/label/421/ The tarball you'll find there includes the OM-6.7 image, source code, a README and the various resources it uses (.so's, libs, fonts, icons, help-files, tutorials, other). This beta is not prepared to be installed anywhere, only to be run from inside the directory where you unpack it (type ./OM_6.7_beta_4 ... ). Plan is to make an installable version once the dust settles some. Theres one thing which needs installing to work however, the music-fonts used in the various editors and factories etc. Theres an .rpm here: http://forge.ircam.fr/p/OM6/downloads/get/omfonts-1.0-1.noarch.rpm.zip - which hopefully takes care of the fonts (theyre also available for manual install in the tarball). The .rpm installs them (4 .ttf-files) inside its own folder /usr/share/fonts/omfonts/* The present version uses JACK for audio + midi I/O, and will try to start a JACK server if its not already running. To get any sound from the usual midi-ports, the linux-port uses libfluidsynth, and attempts to load one of the standard soundfonts into it. However you should be able to route midi-output to any client you prefer (with jack-midi support). Thanks for any feedback. Cheers, -anders

11 years, 8 months

list all ALSA-devices (including virtual ones)

by IOhannes m zmoelnig

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA256 hi all, what's the proper way to list all alsa-devices, including those (plugin, slave,...) defined in asoundrc? e.g. on my desktop i have $ cat ~/.asoundrc pcm.jack { type jack playback_ports { 0 1 } capture_ports { 0 1 } } $ cat /proc/asound/cards 0 [MID ]: HDA-Intel - HDA Intel MID HDA Intel MID at 0xf4ff8000 irq 42 1 [Bt878 ]: Bt87x - Brooktree Bt878 Brooktree Bt878 at 0xf3fff000, irq 17 but aplay will show me: <snip> $ LANG=C aplay -L -l null Discard all samples (playback) or generate zero samples (capture) pulse PulseAudio Sound Server sysdefault:CARD=MID HDA Intel MID, VT1828S Analog Default Audio Device front:CARD=MID,DEV=0 HDA Intel MID, VT1828S Analog Front speakers surround40:CARD=MID,DEV=0 HDA Intel MID, VT1828S Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=MID,DEV=0 HDA Intel MID, VT1828S Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=MID,DEV=0 HDA Intel MID, VT1828S Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=MID,DEV=0 HDA Intel MID, VT1828S Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=MID,DEV=0 HDA Intel MID, VT1828S Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers hdmi:CARD=MID,DEV=0 HDA Intel MID, VT1828S Digital HDMI Audio Output **** List of PLAYBACK Hardware Devices **** card 0: MID [HDA Intel MID], device 0: VT1828S Analog [VT1828S Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: MID [HDA Intel MID], device 2: VT1828S Alt Analog [VT1828S Alt Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: MID [HDA Intel MID], device 3: VT1828S Digital [VT1828S Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 </snip> there is no mention of my 'jack' pcm, *and* there are devices that i don't know where they come from: namely 'null' and 'pulse' (pulseaudio is installed but not running, as confirmed by <snip> $ pgrep pulse $ </snip> so i figure that the extra devices come from system-configuration files in /usr/share/alsa/alsa.conf(.d/*.conf) but why does my "jack" PCM not show up? i know that i can extend my asoundrc to: <snip> pcm.jack { type jack playback_ports { 0 1 } capture_ports { 0 1 } hint { description "use jack as alsa-device" } } </snip> but i don't fully understand why i have to manually add the "hint" (of which i haven't found *any* documentation) in order to make the device show up with char**hints=NULL; snd_device_name_hint(-1, "pcm", (void***)&hints); any clues? fgasmdr IOhannes -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.15 (GNU/Linux) Comment: Using GnuPG with Icedove - http://www.enigmail.net/ iQIcBAEBCAAGBQJSbpcIAAoJELZQGcR/ejb4MdUP/jatXtJ+8OMWVCpWjREoVUqD j73W6ghI16MNHKhOI2DalmZEQM5P3XocBCR0zoG6mzbzIV5TwkvH/tRwjNbvvkoS suinC/7XFB0p0MpaMfzsMmU34ZlvynCalyWL1DBuApxxaFbTeIxHKTQ3/fF5ztpG iM0fyXVKBw7Y0PX4iubl8CeUwmwgvoZ3oV4SqwkgOk8r+cPB/lrkGFl6vrOQxUV0 lQeHdlzPWDSU/QPfuT8OUpN6A4HIjrEMQ2PMQNAOhSBE3N4WJnGxz3La6TeHU3Xu ybKTn70pslyO+bDluwvqygpaMoBRnSfdXW7+JxLfq/0j1ND2FVIRl0RL2aZADPp6 kcq92L+9py5sF48lEVIEFvbTLXEL7uUH1fxPBkbqw2Rde8CY+aPQqDgQvbnDZMTu oN7JEjySB9meJw7RBnAdMQokC6meJDdWJ/pxZuBapvx6+T0m3PaiUZbO5scPSFVt w3j/fA3NipZwZVENJhuiyWO+H1DUbq8lXHYPSSP5YjgyIkzDu36KQigRfSCuZdrb mq/aDXVmC0+j5EWF9lkDYSDwPFj5GqcsuehX86mAS1LLTyL8/IeyjYTVwbN8FjuJ Sb8g+PNFWJDIAo9OrgOEvzYkQ+qbCAf5TBVWKCWe5HdCBlChpXd8yDMKb52dtB7L GCF4kejxZn75aooUxru5 =1pxG -----END PGP SIGNATURE-----

11 years, 8 months

raspberry pi audio in

by Robert Jonsson

Hi guys, As those interested know the raspberry pi lacks an audio input. Now, I came across this some days ago: http://www.noiseisgood.co.nz/?p=365 Apparently it is possible to wire an i2c adc/dac to the raspberry pi and utilize some already available alsa driver. Just thought I would share and ask if anyone has tried anything similar? Regards, Robert

11 years, 8 months

Looking for library to categorise audio

by Philipp Überbacher

Hi there, I'm looking for a library that I could use to categorise audio. Relatively simple categories would be sufficient, for example silence, music, speech, noise. It would be preferable if I could use it from a general purpose language over a specialised one (rather java or C than pd). Do you know or even have experience with such a library? Thanks in advance, Philipp

11 years, 8 months

Gain in dB

by André Garnier Coutinho

Hey guys, Which definition do you use when you want to do a gain control in dB? GdB = 20log(Out/In) ?

11 years, 8 months

LV2 Oscilloscope

by Aurélien Leblond

>> problem is get_supplied_options() returns 0 options when called from the UI. >> > get_supplied_options() should work fine both in UIs and Plugins. My guess > is the host only gives the sampleRate option to the Plugin and not the UI. > Which host are you working with? I'm using Ingen - I sent a ticket to David Robillard onto http://dev.drobilla.net/timeline and will wait for his advise. In the meant time I'll try and debug Ingen itself and see if I can create a patch. > Also, you can compile the UI with a -DLVTK_DEBUG=1, and the console will > print the validation status for any features passed in by the host. > > You could also from the UI send a patch get message and make sure the > Plugin knows how to respond appropriately. Thanks for the ideas - I'll give those a try. Aurélien

11 years, 8 months

Re: [LAD] html5 audio filters

by Patrick Shirkey

On Mon, October 14, 2013 4:08 am, Markus Seeber wrote: > Hi, > i'm not sure, but you could have a look at what he is doing: > > http://mohayonao.github.io/timbre.js/reverb.html > > That is quite a large JavaScript framework, but maybe the right place to > start? > Looks interesting but it doesn't have support for ogg afaict. However I might be able to do something with the examples. Anyone else have any other suggestions? > > Am 10/13/2013 06:58 PM, schrieb Patrick Shirkey: >> Hi, >> >> Can anyone point me to an example for echo/delay/reverb filters using >> html5 audio? >> >> -- Patrick Shirkey Boost Hardware Ltd

11 years, 8 months

Call for testing: ALSA driver for Fireworks/BeBoB based devices

by Takashi Sakamoto

Hi all, This is a call for testing my ALSA driver for Fireworks/BeBoB based devices. Please test 'snd-fireworks' for Fireworks and 'snd-bebob' for BeBoB if you have some devices listed in the end of this mail. Status: - still under development - Without snd-dice and Clemens' development (I must do this later) Functionality: - playback/capturing (full duplex) with PCM/MIDI interface - hardware metering for some devices with CONTROL interface - switching clock source/digital interface/digital mode with CONTROL interface - print hardware status with PROC interface Note: - Don't use simultaneously 'ALSA PCM/MIDI playback/capture' and 'jackd with Firewire (FFADO) backend'. Both of them try connecting to the device when another is running. - I add much modification into snd-firewire-lib for full duplex synchronization of receive/transmit AMDTP stream. Requirement: - Linux kernel 3.11 or later because of Juju (nickname of Firewire stack) changing its API. - Dynamic Kernel Module Support (DKMS) is reccomended for safely installing/uninstalling (I work with Ubuntu 13.10) Bug report: - report with /proc/asound/cardX/#XXX - please send your experiences to me with the output How to install (DKMS): 1. $ git clone https://github.com/takaswie/snd-firewire-improve.git 2. $ ln -s $(pwd)/snd-firewire-improve/ /usr/src/alsa-firewire-3.11 (superuser) 3. $ dkms install snd-firewire/3.11 (superuser) How to uninstall (DKMS): 1. $ modprobe -r snd-bebob snd-fireworks snd-firewire-lib (superuser) 2. $ dkms remove ans-firewire/3.11 --all (superuser) 3. $ rm /usr/src/alsa-firewire-3.11 (superuser) 4. $ rm snd-firewire-improve How to install (Manual): 1. $ git clone https://github.com/takaswie/snd-firewire-improve.git 2. $ cd snd-firewire-improve 3. $ make 4, backup system snd-firewire-lib/snd-firewire-speakers/snd-isight (superuser) 5. install snd-firewire-lib/snd-firewire-speakers/snd-isight/snd-fireworks/snd-bebob (superuser) 6. depmod -a (superuser) How to uninstall (Manual) 1. modprobe -r snd-firewire-lib snd-firewire-speakers snd-isight snd-fireworks snd-bebob (superuser) 2. remove snd-firewire-lib/snd-firewire-speakers/snd-isight/snd-fireworks/snd-bebob (superuser) 3. recover snd-firewire-lib/snd-firewire-speakers/snd-isight (superuser) 4. depmod -a (superuser) Confirmed to work: - AudioFire4 - AudioFirePre8 - Ozonic - Firewire Solo - Firewire Audiophile - Firewire 410 == Fireworks based devices [Echo Audio] AudioFire2 AudioFire4 AudioFirePre8 AudioFire8 (till 2009) AudioFire8 (since 2009) AudioFire12 [Gibson] RIP [Mackie] Onyx 400F Onyx 1200F == BeBoB based devices [Yamaha] GO44 GO46 [M-Audio] (to control mixer channels please use FFADO upstream) Ozonic Firewire 410 Firewire Audiophile Firewire Solo NRV10 ProFireLightbridge [Focusrite] SaffirePro 26 I/O SaffirePro 10 I/O Saffire(LE) [Edirol] FA-66 FA-101 [TerraTecElectronic GmbH] Phase88FW PhaseX24FW [PreSonus] FireBox FirePod [Mackie] OnyxFirewire [Tascam] IF-FW/DM [Behringer] X32 [ApogeeElectronics] Rosetta200 [ESI] Quatafire610 Regards Takashi Sakamoto o-takashi(a)sakamocchi.jp

11 years, 8 months

LV2 Oscilloscope

by Aurélien Leblond

> I would just bypass the LV2 host altogether and use shared memory to get > the audio data to the GUI. > ...and use the same mechanism to send the sample rate. That's through Instance-Access right? That's exactly what I want to avoid - this plugin is meant to be used in Ingen and Ingen doesn't support this. > LV2 UIs can get the sample-rate using the options extension. > Hosts will set the appropriate option lv2param:sampleRate which the UI > can get on initialization. I tried that but without success. I'm using the LVTK library to develop this plugin. My approach is to: - use the get_supplied_options() method to access the host supplied options - iterate until I find SampleRate problem is get_supplied_options() returns 0 options when called from the UI. Any advice? Aurélien

11 years, 8 months

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Linux-audio-dev