Linux-audio-dev December 2010

linux-audio-dev@lists.linuxaudio.org

65 participants
48 discussions

by cal

Does anyone have any experience with speed of traversal through a boost multi index container? I'm pondering their use to manage notes currently in play, eg indexed by midi channel ordered by midi event time/frame stamp. cheers, Cal

14 years

Program design: C++ -> SHM -> Python

by Harry Van Haaren

Hi, I've been trying to come up with a nice program architecture for a live performance tool (Audio looping etc), and I've kind of hit a wall: Input will be taken via OSC, the "engine" will be written in C++, and the GUI is up in the air. I've written most of the engine, (working to a degree needs some bugfixes), and now I've started implementing the GUI in the same binary. Ie: its all compiled together, double-click it and it shows on screen & loads JACK client. The GUI code has a nasty habit of segfaulting.. which is also killing the engine. That's a no-go for live performance. The engine is rock solid stable. So its the fact that there's the GUI thread running around that's segfault-ing things. So I'm wondering if it feasible to keep the audio/other data in SHared Memory, and then write the GUI in Python reading from the same memory? Is this concidered "ugly" design? I have no experience with SHM, so I thought I'd ask. The other option I was concidering is writing the front-end GUI part using only info obtained from OSC, but that would exclude the waveforms of the audio, and lots of other nice features... Help, advice, laughter etc welcomed :-) -Harry

14 years

Simple sound server design

by Olivier Guilyardi

Hello everyone, I am trying to understand how a simple sound server could be implemented. I will not necessarily develop this, but I'm trying to clarify my ideas. As in JACK, it would allow clients to register, and their process callback to be called with input and output buffers of a fixed size. The server would then mix all output data provided by clients and pass the result to the audio hardware. It would also read audio input from the hardware and dispatch it to the clients. There wouldn't be any ports, routing, etc.. as provided by JACK. The main purpose of a such server would be to allow several applications to record and play audio, without them acquiring exclusive access the audio hardware. In this regard it's similar to PulseAudio and many others. The server itself could have a realtime thread for accessing audio. Therefore, for a proof of concept, it could be developed on top of JACK. However, none of the client could run in realtime: this is a given of my problem. The clients would be standard applications, with very limited privileges. They wouldn't be able to increase their own thread priorities at all. Each client would run as an separated process. The only solution that came to my mind so far is to have the clients communicate with the server through shared memory. For each client, a shared memory region would be allocated, consisting of one lock-free ringbuffer for input, another for output, as well as a shared semaphore for server-to-client signaling. At each cycle, the server would read and write audio data from/to the ringbuffers of each registered clients, and then call sem_post() on all shared semaphores. A client side library would handle all client registering details, as well as thread creation. It would then sem_wait(), and when awaken, read from the input ringbuffer, call the client process callback with I/O buffers, and write to the output ringbuffer. Does this design sound good to you? Do you think it could achieve reliable I/O, and reasonable latency? Keeping latency as low as possible, what do you advise for the size of the ringbuffers? -- Olivier

14 years

Identify Peek in Drum Samples

by Oliver Jaun

Hello I bought the natural drum samples (http://www.naturaldrum.com/). It contains WAVs and presets for Kontakt and Halion. Now I'd like to create some gigasampler files in order to use it with linuxsampler. The documentation of the natural drum sample library is quite good. The only thing missing is the "loudness" of each sample in order to map each sample to a velocity level from 0-127. What would you recommend in order to calculate the "peek" of each drum sample automatically? Is there a library which could do this? I would also be happy with a command line tool like this: $ peek bla.wav Peek value: 12345 I could then write a C++-App using libgig. Any ideas? Libraries? Algorithms? Thanks! Oliver

14 years

seeking "fresh" way to process/shape human whisper in real-time

by ico＠vt.edu

Hi all, I've been battling a kind of a dsp-writer's-block as of late. Namely, I am dealing with a project where (at least as of right now) I would like to explore human whisper and its percussive/rhythmic power. This would take place in an ensemble of "voices." I am also looking to combine whisper with some sort of DSP. Obviously vocoder comes as one of the obvious choices but it sounds IMHO cliche and as a result I would like to avoid it as much as possible (unless I can somehow come up with a cool spin on it which I haven't yet). I also tried amp mod, additive, filtering, etc., but none of these struck me as something interesting. I do think delays will be fine in terms of "punctuating" the overall pattern but I think this should take place at the end of the DSP chain. Granular synthesis is also a consideration but I've done so much of it over the past years I am hoping to do something different. So, as of right now I have: 1) whisper 2) ??? 3) delays 4) profit! :-) Given the mental constipation I have been battling particularly over the past couple of days, I wanted to turn to you my fellow LA* enthusiasts for some thoughts/ideas/inspiration. Your help would be most appreciated and I will gladly credit your ideas in the final piece. Many thanks! Ivica Ico Bukvic, D.M.A. Composition, Music Technology Director, DISIS Interactive Sound and Intermedia Studio Assistant Co-Director, CCTAD CHCI, CS, and Art (by courtesy) Virginia Tech Department of Music Blacksburg, VA 24061-0240 (540) 231-6139 (540) 231-5034 (fax) ico.bukvic.net

14 years

Fwd: Qtest Mobile App Port Contest for Qt and KDE applications

by Niels Mayer

If you need something to push you over the edge and port your existing Qt/KDE music-making or multimedia app to the N900 running Symbian, Maemo, or Meego: http://qt-apps.org/news/?id=340 (see below). Some ideas (please?): http://sv1.sourceforge.net/ == http://www.sonicvisualiser.org/ http://kmid2.sourceforge.net http://kmetronome.sourceforge.net http://kmidimon.sourceforge.net http://vmpk.sourceforge.net/ http://qtractor.sourceforge.net http://qmidictl.sourceforge.net http://qmidinet.sourceforge.net http://qjackctl.sourceforge.net/ .................... Win 10.000,- EUR at the "Qtest Mobile App Port" Published: Dec 20 2010 Qtest Mobile App Port Contest for Qt and KDE applications Welcome to the Qtest Mobile App Port! As developers of applications using Qt, you already know how great it is to work with - but how about on mobile platforms, such as Symbian and MeeGo? How would you like to take that step you have been wanting to take, but not been able to justify: Take your application from the desktop and bring it into the hand-held world via the Ovi store. Let this contest be the justification, with the possibility of a new phone or even 10,000 euros waiting at the end. Dates: The contest starts on 20th of December, 2010, and runs till 28th of February. The 31st of December is important for you if you wish to take part in the Early Bird competition. If you do no win, you will still take part in the main competition, and will be allowed to continue your work and submit new versions to the Ovi Store. The 28th of February is the deadline for taking part in the main competition. Developer Sprint: There will be a sponsored developer sprint organized together with the KDE e.V. during the competition. The travel and stay can be paid for if you do not have the budget yourself. Further details will be made public at a later time, and all participants will be notified of this information via email. Judging and prizes: The Qtest Mobile App Port is evaluated by a panel of judges which will be announced in the next few days. The jury will pic 5 winners at 31th of December as the early bird winners. Every winner gets a free N900 phone. The main competition first prize is EUR 10,000, which will be awarded to the application which the judges find to be the best ported application. The second to sixth price will be another 5 N900 phones. And, finally: Everybody who takes part in the competition will be awarded a gift bag, with a T-shirt and other merchandise. Eligibility: To be able to take part in the contest, the ported application must be submitted for Ovi Store signing by one of the two deadlines: - Early bird entries must be submitted by December 31st - Standard entries must be submitted by February 28th You also have to submit your application to the "Mobile Contest" category on Qt-Apps.org or MeeGo-Central.org You can submit your application to the Ovi Store as many times as you wish during the competition. This allows you to get feedback from the public on your software. It´s possible to submit new or existing KDE/Qt applications So have fun and good luck everybody! ....................... Niels http://nielsmayer.com

14 years

Best wishes for the Holidays

by Ivica Ico Bukvic

Dear Friends, On behalf of Linuxaudio international consortium (http://linuxaudio.org), DISIS (http://disismusic.vt.edu), and L2Ork (http://l2ork.music.vt.edu), please allow me to use this opportunity to wish you very best for the Holidays. May you have many more seasons of merry music making using an ever-growing array of formidable FOSS tools and solutions! Best wishes, Ivica Ico Bukvic, D.M.A. Composition, Music Technology Director, DISIS Interactive Sound & Intermedia Studio Director, L2Ork Linux Laptop Orchestra Assistant Co-Director, CCTAD Virginia Tech Department of Music Blacksburg, VA 24061-0240 (540) 231-6139 (540) 231-5034 (fax) ico.bukvic.net

14 years

job posting: ALSA/gstreamer/linux/C++/FOSS Indiana, USA

by Paul Davis

I'm forwarding a job posting sent to me by the recruiter. I'd like to do this myself, but the time and location make it infeasible. Apparently, there are some options for remote work but I'm not clear what they are. Talk to the recruiter to find out more. --p ---------------------------------------------------------------------------------------------- Tom Gugger Independent Recruiter tgugger(a)bex.net Linux Audio/ Contract/ Immediate/ IN Skills: linux ubuntu embedded alsa gstreamer cm synergy c++ This is an eight month contract in Kokomo, Indiana. If you fit the job description below, email your resume to tgugger(a)bex.net. Make sure your resume shows the required skills. No ALSA type experience, no job. ALSA = Linux Audio Package Experience Duties in this role include: - Design SW Architecture for Linux and audio subsystem - Detailed task planning and estimating - Linux package management, integration and direction THIS IS A 6-8 MONTH CONTRACT OPPORTUNITY Requirements - Linux Audio Package experience (ALSA source/sink, GStreamer architecture) - Linux Kernel understanding - Ability to reconfigure and rebuild kernel and kernel modules - C++ Object Oriented design, analysis - Linux Driver architecture understanding - It is preferable that the person has contributed to Open Source development (FOSS project) - CM experience required. Synergy experience preferred.

14 years

Re: [LAD] Meego pulseaudio "compliance" and "enforcement" (was Re: [Meego-handset] Enabling Speakerphone)

by Niels Mayer

Marco -- many thanks for your reply -- I've taken the liberty of CC'ing this informative reply to the Linux-Audio-Developers list (your CC was not posted as you're not a subscriber: http://lists.linuxaudio.org/pipermail/linux-audio-dev/2010-December/thread.… ). The list comprises a good number of people with expertise in both pulseaudio; hopefully the Jack sound server authors, including Paul Davis, will be willing to publicly share their perspectives on the issues raised regarding the role of pulseaudio on a handset and Linux audio performance/latency/efficiency issues. Here's a link to Marco's original post to meego-handset mailing list, forwarded below: http://lists.meego.com/pipermail/meego-handset/2010-December/000090.html ---------- Forwarded message ---------- From: Marco Ballesio <gibrovacco> Date: Sat, Dec 18, 2010 at 4:42 AM Subject: Re: Meego pulseaudio "compliance" and "enforcement" (was Re: [Meego-handset] Enabling Speakerphone) To: Niels Mayer <nielsmayer> Cc: meego-handset(a)lists.meego.com, Linux Audio Developers <linux-audio-dev(a)lists.linuxaudio.org> [...] Hi, sorry for the late reply but, whew, this was really long, definitely more than what my poor brain can handle in a single bunch ;). Maybe we could split this into a "Resource Policy Framework" and "Pulseaudio on Embedded" couple of threads. As a general comment on PA, it may appear odd but when I started working with it I shared 100% of your considerations. Working with it (not ON it), I started realising that, ok, it may not be perfect (and which software actually is?) but it brings definitely more benefits than troubles to the system. See my notes below for more details. ..snip.. > > Hopefully there's more than just source-code to describe the policies > and enforcement. Maybe you've read in some other of my posts here and there, I'm trying to get a Resource Policy Framework wiki somewhere. In the meanwhile, documents are scarce but present: - The MeeGo conference presentation: http://conference2010.meego.com/session/policy-framework-flexible-way-orche… Does not say too much, but it comes with some slides and many "mmmmh" (keepalive signals). - The resource policy application developer's manual: http://wiki.meego.com/images/Meego-policy-framework-developer-guide.pdf Mysteriously, it's not linked or referred to anywhere.. at least, now it's on this thread. - The source code: it may twist more than one nose but, at the end, I always like to check in the sources before/after Googling. Documents may be obsolete (who knows?) or expose a partial view. Code can't ;). Not that I want to say with this that we don't need more documents about the Resource Policy Framework... > Is there a design document stating the > "organizational" or "legal" or "human" reasons for said policies and > enforcements? My pointers of above may begin to satiate your hunger (appetisers?). > > And, as a developer, where is the documentation for the appropriate > layer to use, the appropriate API, etc -- e.g. for the original > question that started this thread -- how to switch on/off speakers, > FM-radio transmitter, etc on handset. ( cf my original reply > http://lists.meego.com/pipermail/meego-handset/2010-December/000066.html > ) Unfortunately, this is still in the "dark side" of the documentation, which is currently application-centric. Hopefully one day we'll have an "Adaptation developer's guide", in the meanwhile, a good starting point may be checking the sources, the N900/Meego ruleset and posting questions here. If it's not enough, we may set up an IRC channel (if there's enough interest)... > > Something high-level like the following Maemo document would be very > helpful -- but I have been unsuccessful finding such documentation for > Meego: http://wiki.maemo.org/Documentation/Maemo_5_Developer_Guide/Architecture/Mu… See my notes above about the project wiki. Your pointers are definitely good hints, thanks. > > Also, why not use ALSA use-case-manager: > http://opensource.wolfsonmicro.com/node/14 > UCM appears to be scheduled for other distros "handset" work: > https://wiki.ubuntu.com/Specs/N/ARM/AlsaScenarioManager this project covers the audio-specific cases: for what I understood it might very well be used as an ALSA enforcement point in the Framework so, more than compete with Resource Policy it would be complementary in case you don't really want/need to use a sound server. BTW, I could not find so much info, maybe you can help me to retrieve: - more than just source-code to describe how it works. - the developer's documentation for the appropriate layer to use, etc. - something like: http://wiki.maemo.org/Documentation/Maemo_5_Developer_Guide/Architecture/Mu… ..snip.. now, the pulseaudio cases. First of all, please note I'm not the best advocate for it, btw my 0.05 €. Posting some extracts of your comments to the PA mailing list will definitely give you a better insight for the internals of the server. > Regarding my old nemesis, pulseaudio ( http://tinyurl.com/2976vu6 > == http://old.nabble.com/uninstall-pulseaudio-to-increase-audio-app-stability-… > ), I think the only "elegant" thing about pulseaudio is that it's > bluetooth handling works for those that care about bluetooth; given > the number of bug-reports and incompatibilities pulseaudio generates > in different distros, I'm not sure "elegant" is the right word.... And actually "elegant" was not referred to pulseaudio, but to the way its ports can handle audio routing instead of using ALSA (they bring better synchronisation). Yes, the sound server has shown instabilities in some cases and I personally used to uninstall it on my Ubuntu box immediately after the upgrades. Recently but many improvements have been made at a point that, on a PC, I'm nowadays not seeing it anymore jumping to a steady 30% of the CPU or completely jamming any audio output in the middle of a movie. Another _personal_ feeling I had is that pulseaudio was pretty hard to configure for a generic HW (like distros tend to do), but works pretty well after being tuned for a given one (like 95% of embedded products are). Very often the root issue may have been a wrong configuration for your system. > > From my position, as a multimedia/ALSA/linux-audio developer, having > to go through pulseaudio sounds like an all-around bad idea, and to > have "enforcement" or "compliance" attached makes it sound even > worse. Tell me it ain't so! I can agree from a kernel/ALSA pov (everything in userland appears to be less efficient than any kernel drivers ;) ), I strongly disagree from the system architecture's one. Some points you should consider: - Audio outputs frequency tuning: in the real word a crappy speaker costs less than a good one (strange?) so, unless you've a very high target price, you'll usually have to take the maximum out of garbage by "massaging" the output spectrum in order to get an adequate overall transfer function. And this must work with all of the applications -even the ones coming from third parties on which you don't have the bare minimum control on- and different analog outputs (front, rear, bt, ...). Not that pulseaudio comes with these features out-of-the-box, but it's quite easy to plug them in. - Audio inputs: the dual of audio outputs. - Mixing: it's possible that users will want more than one application accessing the audio output at the same time (e.g. navigator and car radio) or that they'll want to switch bw two different applications all of a sudden (e.g. radio and ringtone). In all these cases they'll want a good mixing and no sudden power steps when the ringtone app does not know about the volume level used for the radio. - Acoustic delay estimation: sitting in the middle of pulseaudio (running as a real-time thread) it's easier to write a proper AEC. Back to 2004 I've tried to grant, with a TI aic31, constant latencies for both LEC and AEC by just using ALSA and the application... It took a really long time to work. - It may not appear so, but it's energy-efficient (see my comments below if you just jumped out of your chair). Now, I don't mean that those things can be handled only by pulseaudio. Actually developers could write their own sound server, or use an already available one, to do all of this, but.. - How long would it take to be better than pulseaudio? (e.g. wrt my points above). - Can it grant the same solid user base and portability across embedded HW? - Could it grant the same level of contribution from companies with solid experience on the subject? - Are there architectural flaws for which pulseaudio couldn't effectively handle these cases? If yes, has anybody tried to discuss about them on the PA mailing list? > > There is a growing class of applications that do not want or need > pulseaudio around -- those using http://jackaudio.org/ . When the > jack audio server launches, the first thing it does it use dbus to > disable pulseaudio. Is that also non compliant? I've no religious issues against jack and actually I don't have that much experience with it, so I'd like to know more about jack on embedded. I'd like to know, for instance, how many embedded, low-power devices are already using it and with which degree of success. Also it would be great to know if anybody has interfaced it with a cellular modem to handle Circuit-switched cellular calls, and if the device has actually been certified for such a service in any countries. > > It seems inappropriate to preclude an entire class of application -- > real-time "pro" audio and video apps that utilize the Jack audio > server to provide low-latency, tightly synchronized audio -- as needed > for modern multimedia creation and playback. Perhaps such applications > are a stretch for an OMAP3-class device, but given the many > audio/media apps listed in http://omappedia.org/wiki/PEAP_Projects , > clearly OMAP4 and beyond might not be, even on a puny "handset." Of > course, those making such audio apps might sidestep pulseaudio > compliance/latency/inefficiency issues by using > http://opensoundcontrol.org/ and an external DAC ( > http://gregsurges.com/tag/dac/ ). > please correct me if I'm wrong, but if I understood well most of the apps using _exclusively_ jack are aimed to audio/video composition/editing. Now, as we are on the IVI ML I think it's quite a strange use case (even though I must admit I don't know which requirements you have). > Finally, it seems odd that in a "handset" environment, pulseaudio is > an absolute requirement. To me, it is just a waste of batteries, see below (and above). > and a > terrible source of unnecessary context switching and interrupts during > audio playback. well, it runs as real-time priority, so it's executed when it's scheduled. Context switches would not be different for any process (e.g. jack) running with the same scheduling. > It's sort of like being forced to drive around in a > gas-guzzling, oversized sport-utility vehicle with 10 foot tires and 5 > feet of ground clearance -- just to drive to the market or work on a > well-paved freeway on a summer's day -- even when one might prefer a > bicycle, motorcycle, sports-car, subway, or whatever tool is best for > the job. funnily, this is EXACTLY the same thing I thought the first time I profiled pulseaudio on the N900, then I found some reasonable explanation (scroll a little below). ..snip.. > Take for example HD audio/video playback -- something where you need a > "sportscar" to get the latency and sample rate associated with the > audio stream while also performing HD decoding (e.g. 16bit/96K audio > is supported on omap3430 per > http://and-developers.com/device_information ). Pulseaudio typically > operates at a fixed rate and forces resampling to that rate, causing > an oft perceptible loss in fidelity. Yes, this is a bad aspect, but it's actually possible to change the output bitrate of the server (it depends on the port used). Maybe you could query on the PA mailing list for more knowledge on the subject. > > IMHO what is needed is not a "digital mixer" and resampler and extra > user-space processing before going into the kernel and out to the > sound hardware via ALSA drivers. We certainly need "use case > management" and the notion of a digital patch bay, and some way of > smoothly mixing between sounds, glitch free switching of sample rates > at the ALSA level, and then choosing the direct path to hardware > that'll best handle the "main audio task" we're doing -- e.g. playing > music, watching a movie, making a phone call, or just using the screen > UI. What isn't needed is "desktop networked audio" capabilities of > pulseaudio, or any extra user-space streaming/mixing/resampling of > signals. so we'll end up writing our own sound server or using a different one, won't we? As an alternative, we may have to heavily modify ALSA to suit our needs, and we'll not have yet covered all of the needed features. > > The inefficiencies introduced by pulseaudio is evidenced by the > n900/meego-1.1 handset, which cannot maintain audio synchronization > during audio or video playback if even the slightest trace of > something else is going on, including a wireless network scan, or even > just cursoring around in a networked terminal (which goes through the > wireless stack in my setup). Personally, I would not say that MultiMedia with MeeGo on the N900 is already at the same quality of Maemo 5. Lots of tuning is imho still missing. > > On Maemo/n900 note what happens when playing back an Mp3 -- pulseaudio > consumes twice the resources at "high priority" of the decoding > process: You must consider that here there many algorithms involved that users/applications don't even know about, like the spectrum optimisations I've been mentioning before. If you could run an oprofile session over your tests you'd see that the actual CPU attributable to PA is pretty lower than your figures. ..snip.. > I'm sure a simple experiment could determine exactly the "effect" of pulseaudio: > > Play the same playlist until the battery wears out using the same > software player outputting first to pulseaudio (pref not through > ALSA's pulseaudio driver because that wouldn't be "fair" to go through > the kernel twice) then play the same through ALSA "dmix" interface, > just to emulate pulseaudio's mixing/resampling functionality. As written above, PA is not only resampling and mixing here.. imho if the user/the developer didn't have to adjust the system to work with PA in order to achieve this, it's a success. The CPU usage comes from the fact we're actually running something we need. > I would > imagine the pure-ALSA solution would pass the "energizer bunny' test > for more hours and far fewer kernel/userspace context switches. Sure it would, but with a lower set of features (and an higher cost for the audio system). It's up to the device price target/customer's pockets and needs to decide what's better. > Although a realtime userspace process like pulseaudio can help deliver > stable audio in a multicore environment -- it may end starving out > other processes on a slower uniprocessor. Which is why I believe a > pulse-audio-free solution should be available and still be "compliant" > on low-end systems. I would agree in some cases, that is when the system conditions don't require it to run multi-application use cases and the price tag for the audio system is irrelevant wrt the audio quality. Note for the reader: if you've reached this point it means you're really interested about pulseaudio on embedded devices :D. Regards > > The one place where pulseaudio is currently helpful is in hooking up > bluetooth devices. But ALSA has it's own bluetooth layer as well, and > other architectures for audio shouldn't be precluded: > http://bluetooth-alsa.sourceforge.net/future.html or Phonon > (http://pulseaudio.org/wiki/KDE ). > > -- Niels > http://nielsmayer.com >

14 years

slaving to jack_start/stop

by Julien Claassen

Hello! If I want to have a simple jack enabled program react to transport control in JACK, what do I have to do? The basics: * the program can start (it will play always from the same point (so no need for relocating, etc.) * the program can stop * I also have an interactive command to make it start. I guess in this case: Just let it start and leave the transport master untouched. Can I easily add the following: When the program is running, due to the interactive start command and external jack_transport play/start comes along, just start over playing? Kind regards and thanks Julien -------- Music was my first love and it will be my last (John Miles) ======== FIND MY WEB-PROJECT AT: ======== http://ltsb.sourceforge.net the Linux TextBased Studio guide ======= AND MY PERSONAL PAGES AT: ======= http://www.juliencoder.de

14 years

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Linux-audio-dev December 2010