Linux-audio-dev

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7516 discussions

LMMS Trippleoscillator, LV2/linuxVST-version possitiblity?

by Aurélien Leblond

> My question is, quite simply, since LMMS is FLOSS, would making > Trippleoscilator + perhaps other instruments in LMMS into plugins > (lv2/linuxVST) be a realistic possibility? Bear with me, I'm no developer > but merely a musician, so I really don't know what I'm asking/talking > about. But I have a firm belief that a plugin-version of these instruments, > usable in for example Ardour3 when that arrives, would make a great > addition to the Linux-musician community. Technically it should be doable, but one would need to go in the code of LMMS to check it out. Sometime the code is almost usable directly in plugins. I'm working on porting the internal modules from AlsaModularSynth into LV2 plugins at the moment. Once I'm finished with that, I could take a look into it. (http://sourceforge.net/projects/avwlv2/) > (ps. Is there anything similar already existing perhaps? ds.) You should check AlsaModularSynth, coupled with Seq24... The instrument you described look like a synth with 3 oscillators (duuuuh, that would explain the name!). AMS comes with great examples, and building a Synth with 3 VCO (Oscillators) should be "pretty" easy. Yoshimi comes to mind as well (although it is way more complicated). Oh, and if you have some courage, feel free to try my plugins with Ingen :D P.S: I forwarded your mail to the LAU list. Some users there might have other ideas for similar synths. Hope that helps, Aurélien

13 years, 10 months

Denormals / subnormals (again)

by Carl Hetherington

Hi all, I have been looking into ye olde denormal problem a little, lately. Particularly with respect to Ardour and plugins. I've assembled what I believe to be a coherent statement of what is going on, but I'd very much appreciate any corrections and clarifications that anyone can offer. Here's how it seems to me: If you compile your code (e.g. a plugin) without -msse and -mfpmath=sse on the GCC command line you get *no* protection from denormals from the CPU. If they occur in your code, they will be very much slower than normal floating point numbers (~49 times slower on my Core 2 Duo, ~7 times slower on a Core i3). As far as I can see, it does not matter that Ardour has been built with those flags: if a plugin has not, you have no protection. If you compile your code with -msse and -mfpmath=sse, you have Ardour's protection from denormals. If the user's CPU supports it, you get there is no significant slowdown with denormals using this mode. However CPU support is "some later processors with SSE2", according to Intel. The problem, I guess, is that we cannot really distribute plugins with SSE instructions, otherwise we do not support people with older CPUs. In this case, I think the plugin code must avoid denormals, otherwise there will be a significant performance hit if they arise. I've been testing behaviour using a very dumb program which you can get from http://carlh.net/software/denormals.tar.gz I've also been testing plugins using a primitive torture tester that you can get from http://carlh.net/software/ or on github via git@github.com:cth103/plugin-torture.git Any comments? Best Carl

13 years, 10 months

USB-Audio and XHCI (USB 3)

by Aurélien Leblond

Hi everyone, I was wondering if anyone managed to run a USB 2 soundcard on a USB3 (XHCI) port? I'm running the kernel 3.2 rc7. With the M-Audio Fast Track Ultra (USB 2): The output of dmesg is this: [ 475.094595] usb 3-1: new high-speed USB device number 2 using xhci_hcd [ 475.108605] usb 3-1: config 1 interface 3 altsetting 0 bulk endpoint 0x7 has invalid maxpacket 8 [ 475.108615] usb 3-1: config 1 interface 3 altsetting 0 bulk endpoint 0x87 has invalid maxpacket 8 [ 475.108958] xhci_hcd 0000:04:00.0: WARN: short transfer on control ep [ 475.109328] xhci_hcd 0000:04:00.0: WARN: short transfer on control ep [ 475.109879] xhci_hcd 0000:04:00.0: WARN: short transfer on control ep [ 475.110564] xhci_hcd 0000:04:00.0: ERROR: unexpected command completion code 0x11. [ 475.110585] usb 3-1: can't set config #1, error -22 The snd-usb-audio module is not probed automatically, and if I modprobe it manually, no sound card is detected. The error message is described in this bug: https://bugzilla.novell.com/show_bug.cgi?id=679944 but if I understand well, kernel wise, everything is "ok" and the issue lies in Alsa The output of lsusb is: Bus 001 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Bus 002 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Bus 003 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Bus 004 Device 001: ID 1d6b:0003 Linux Foundation 3.0 root hub Bus 001 Device 002: ID 8087:0020 Intel Corp. Integrated Rate Matching Hub Bus 002 Device 002: ID 8087:0020 Intel Corp. Integrated Rate Matching Hub Bus 001 Device 003: ID 050d:0307 Belkin Components Bus 002 Device 003: ID 0408:1af1 Quanta Computer, Inc. Bus 001 Device 004: ID 046d:c51b Logitech, Inc. V220 Cordless Optical Mouse for Notebooks Bus 003 Device 002: ID 0763:2080 Midiman M-Audio RunTime DFU The Edirol UA-25Ex (USB 1) works fine, I get a few warnings in dmesg but the sound card is detected and functional... Let me know if I should provide any more info, or outputs, anything that can help. Kind Regards, Aurélien

13 years, 10 months

segfaulting using my ringbuffer queue, stuck!

by Iain Duncan

Hi folks, wondering if anyone might be able to point me at the way to sort this out, my latest queue is segfaulting when written to or read from, while the rest ( which are supposed to be identical except for message type ) are working great, and have been tested with full stack runs fine. I'm banging my head on the desk at this point trying to find the difference, but perhaps others have seen similar behaviour? I've made a template class for a queue, that internally uses a jack ringbuffer. I have four of them, some of my data message struct, one for a csound note message struct, and a new one for my raw midi message struct which looks this this: struct MidiMessage { char status; char data_1; char data_2; int time; // time in samples when midi message arrived }; I instantiate them before anything else and pass them into the components that need them in their constructors MessageQueue<DataMessage> *toEngineDataQueue = new MessageQueue<DataMessage>(); MessageQueue<DataMessage> *fromEngineDataQueue = new MessageQueue<DataMessage>(); MessageQueue<NoteMessage> *toEngineNoteQueue = new MessageQueue<NoteMessage>(); MessageQueue<MidiMessage> *fromEngineMidiQueue = new MessageQueue<MidiMessage>(); All instantiation is working fine, app starts up, and the first three queues are working. As soon as I either write to or read from the midi queue, I segfault. Not sure how to debug this, hints welcome! Below is the cue code in case anyone wants to look at it. I can't see anything wrong, but maybe I've been doing something wrong and just gotten lucky so far?? thanks Iain template <class Type> class MessageQueue { private: // pointer to the ring buffer the ring buffer jack_ringbuffer_t *mRingBuffer; int mQueueLength; public: MessageQueue(); ~MessageQueue(); // put a msg on the queue, returns 0 or error code void push( Type msg ); // store message in msg, returns true if message bool tryPop( Type *msg ); }; template <class Type> MessageQueue<Type>::MessageQueue(){ mQueueLength = DEFAULT_QUEUE_LENGTH; // create our ringbuffer, sized by Type mRingBuffer = jack_ringbuffer_create( mQueueLength * sizeof(Type) ); // lock the buffer into memory, this is *NOT* realtime safe int errorLocking = jack_ringbuffer_mlock(mRingBuffer); if( errorLocking ){ std::cout << "MessageQueue - Error locking memory when creating ringbuffer\n"; // XXX raise an exception or something?? how do we fail here?? } } template <class Type> MessageQueue<Type>::~MessageQueue(){ cout << "MessageQueue destructor\n"; // free the memory allocated for the ring buffer ack_ringbuffer_free( mRingBuffer ); } template <class Type> void MessageQueue<Type>::push( Type msg ){ // write to the ring buffer, converting Type to a string unsigned int written = jack_ringbuffer_write( mRingBuffer, (const char *) &msg , sizeof(Type) ); // XXX: what to do if it fails anyway?? if( written < sizeof(Type) ){ cout << "Error, unable to write full message to ring buffer\n"; // do something else here yo! } } // if a message is on the queue, get it // returns True if it got a message template <class Type> bool MessageQueue<Type>::tryPop( Type *msgBuf ){ // if there is a message on the ring buffer, copy contents into msg if( jack_ringbuffer_read_space( mRingBuffer) >= sizeof(Type) ){ jack_ringbuffer_read( mRingBuffer, (char *)msgBuf, sizeof(Type) ); // return True because a msg was read return 1; }else{ // return False, no msg read return 0; } }

13 years, 10 months

handling midi input in a jack app?

by Iain Duncan

Hey folks, what is the easiest way to deal with midi input in a jack app? I'm confused by the difference in jack midi and alsa midi, because I have two midi inputs, one is a usb input, so it appears at a low level as an alsa device, but the other is the midi input on a firewire unit, and it appears as a jack midi device. I'd like to make sure that whatever I do is easy to port to other systems. Does it make sense to use portmidi or rtmidi to get input or should I stick to the jack api entirely? thanks Iain

13 years, 10 months

Scopes & RMS

by Harry van Haaren

Hey all, I've been writing a scope the last while, and I'm intrested in how other people have approached plotting the data. Currently I'm taking every 50th sample, and drawing a line from the previous sample to there, and so on. Not particulary neat. So is there a resample, or smoothing of the samples, or how does one plot a "smooth" waveform like Ardour / QTractor? My other question is about RMS, is calculating it in its literal sense best? Or perhaps only taking every 5th sample? Resampling the signal from 44.1 or 48k to say 11025? I'm not really sure which way to go. Cheers, -Harry

13 years, 10 months

Replaygain for video?

by Philipp

Hi there, I could use some advise. You may or may not heard of replaygain. It's reasonably widely used in consumer audio, but sometimes I wish it was available for video as well. By this I mean I wish it was available for the audio part of the video. Well, I need a programming project for a university course and this is just one of my ideas that I want to propose to my teacher and prospective teammates. In order to do this I'd like to narrow it down a bit further and especially want to find out whether I have the right idea of how it can be achieved. Scanning/tagging Since replaygain works on whole audio files I think I need to extract the whole audio track from the container. How easily this can be achieved I don't know. After that, the scanning process should work as with any audio file. Afterwards the calculated replaygain values have to be added to the metadata of the file. I have no idea how hard it is to add new metadata fields to video formats. Playback Video players need to be aware of those tags, read the metadata and scale the playback volume accordingly. This is probably not hard per se, but there are many players out there. However, I plan to start with a single player, even with a single file format, and go from there. Question 1: Is there anything better than replaygain that should be used instead? Question 2: Which player would be easiest to hack to add such functionality? Could it be a gstreamer plugin? mplayer? Question 3: How much work would it be? The project should be done in C++ if possible, otherwise C. Group size 2-4 Students, all rather new at C/C++ and rather inexperienced in general. Other ideas I have are in short: - A CLI (using readline) connection manager for jack audio/midi and alsa midi that can handle large numbers of ports. More detailed ideas exist thanks to Julien Claasen. - A simple but hopefully sane mplayer GUI - A new GUI for ecasound Another problem I might have is that most students in the course are Windows users, not sure whether I can go solo. Thanks for any advice, regards, Philipp

13 years, 10 months

Debugging environment for lv2 development.

by email.rafa＠gmail.com

Hi everyone. I'm starting to write a simple filter and I want to expose it as an lv2 plugin. My development environment is very simple right now: vim editor, gcc compiler, package the lv2 manually (will write a script for that in a day or two) and then load the plugin in ardour to test it. I've found this setup to be a bit unconfortable because once I load the plugin in ardour I don't know how to get debug information from it (print statements or breakpoints with gdb). What does a lv2 development environment typically looks like? what are you guys using? Thanks! -- Rafael Vega email.rafa(a)gmail.com

13 years, 10 months

Csound 5.15 release

by Victor Lazzarini

Apologies for cross-posting. ====================== We are pleased to announce the release of version 5.15. The sources are on the standard Sourceforge location (https://sourceforge.net/projects/csound/files/csound5/csound5.15/) as both zip and tar.gz Platform packages will follow shortly, and the manual on Friday. ==John ffitch ------------------------------------------------------------------------ Notes for 5.15 ============== New parser has been subjected to a great deal of work. It now has better checking of argument types and use, better diagnostics and increased functionality. We have only reached this stage in the last few days so we judge it prudent to leave the old parser as standard. We would be pleased if more users tried the new and gave the developers feedback. A major reorganisation means that there are many fewer plugins and most opcodes are in the base (about 1250 of them). A side effect of that is that leaving old plugins from an earlier release is a disaster, and so 5.15 will not load earlier plugins. The multicore system is now safe (ie maintains semantics) when zak, channels or table modification are made. New Opcodes: ftab2tab transfers between ftables and t-variables tab2pvs tsig - pvs conversion pvs2tab pvs - tsig conversion cpumeter-- not really new but now available in OSX (EXPERIMENTAL) ftresize and ftresizei allow resizing of existing tables. These will be permanent if the community feel they are useful. minmax opcodes hrtfearly, hrtfreverb opcodes New Gen and Macros: Code to allow GEN49 to be deferred [NB does not seem to work] Modified Opcodes and Gens: socksend and sockrecv no longer uses MTFU check and work on Windows mpulse changed so if next event is at negative time use the absolute value serial opcode now runs on Windows as will as Un*x out, out2, outq, outh, outo outx and out32 are now identical opcodes and will take up to as many arguments as nchnls. This replaces the current remapping of opcodes turnoff2 now polymorphic wrt S and k types (ie accepts instrumnet names) Utilities Bugs fixed: GEN42 fixed jacko: fixed a segfault removing the unused JackSessionID option doppler memory leak fixed transegr fixed in release mode when skipping most of envelope FLPack now agrees with manual max_k now agrees with manual hrtfreverb fixed atsa code now works on Windows in more cases tabmorph bug fixed fixed problem with user-defined opcodes having no outputs Various fixes to * ... */ comments System Changes: Various licence issues sorted Loris is no longer part of the Csound tree Memory leaks fixed If no score is given a dummy that runs for over 100 years is created All score processing takes place in memory without temporary files String memory now expandable and no size limitation #if #else #end now in new parser Adjustments to MIDI file precision in output On OSX move from Coreaudio to AuHAL Multicore now safe for ZAK, Channels and modifying tables New coremidi module Virtual Keyboard improved: 1) Dropdown for choosing base octave (the one that starts with the virtual key mapped to physical key Z). Default value is 5 which is backwards compatible. 2) Shift-X mappings which add two octaves to X mappings for a total of 4 octaves playable from the physical keyboard (starting from selected base octave). 3) Control-N / Control-Shift-N mappings to increment / decrement slider for control N. 4) Mouse wheel now controls sliders. tsig type for vectors tsigs and fsigs allowed as arguments in UDOs API: Minor version upped Internal: Very, very, very many! Dr Victor Lazzarini Senior Lecturer Dept. of Music NUI Maynooth Ireland tel.: +353 1 708 3545 Victor dot Lazzarini AT nuim dot ie

13 years, 10 months

how to store deferred events in a real time audio app?

by Iain Duncan

Hi, I'm sure others have tackled this and have some wisdom to share. My project is principally a monosynth step sequencer. This is nice an simple to do in real time because resolution is very limited and there can be only one note per track. So step sequenced note data is stored in simple multi-dimensional arrays, making reading and writing very easy, and messaging simple between audio and gui threads. However, I would like to add the ability for the user to send a message and have it get executed later, where later gets figured out by the engine ( ie on the top of the next 8 bar phrase ). To do this, I need some way of storing deferred events and having the engine check on each step whether there were any deferred events stored for 'now'. I can think of a few ways to do this, and all of them raise red flags for a real time thread. - I could use a hash table, hashed by time, with a linked list of all the events for that time. The engine looks up the current time and gets all the events. I don't know much about hashing so I'd prob just use Boost, is that a bad idea? - I could make a linked list of all deferred events and iterate through them checking if time is now. There wouldn't be any hashing, but maybe this list would be really big. Anyone have any suggestions for how to safely do the above or some better alternative? thanks! iain

13 years, 10 months

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Linux-audio-dev