Linux-audio-dev

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3 participants
7535 discussions

Re: [LAD] FOSS Ethernet Soundcard - IETF Ethernet AVB

by errordeveloper＠gmail.com

Hi everybody, Fallowing up the long discussion i'm trying to sort of give the information you all seem to be missing. there was a meantion of this, but not too many of you have paid any attention: here you can find an article about the current status of the protocol mess: http://prosoundnewseurope.com/pdf/PSNLive/PSNLive_2009.pdf (page 28) obviously eas50 is good to go, but Ethernet AVB is right thing really. the only thing that it's still work in progress, but many of the proprietary vendors, which already have their own networking solution (like Harman with HiQ-net, the one i can name of top of my head) are involved in AVB stadard deveelopment. The idea of AVB is to bypass the IP layer, which is right thing really. you don't need to assign IPs to your audio nodes, really! in avb you'd just have to select channels that nodes whant to listen to. there is a fair bit of documentation on the ietf.org AVB group's page. but XMOS is looking to be the best point of refference: http://www.xmos.com/news/15-jun-2009/xmos-simplifies-ethernet-avb-implement… is think we should forget everything else and crack on with the XS1 AVB implementation! their XS1 chips seem to be really great, their are basically every innovative and open-source minded. the official toolchain is LLVM-GCC based. you can use C, C++ or their own XC. XC is basically C with some stuff omited (like goto and floats) and XMOS IO stuff added, don't just say WTF, look at it first! you should also watch the videos here: http://www.xmoslinkers.org/conference-online-wf especialy the two about the "XMOS Architecture" and the AVB presentation. some dev-kits are quite expencive, but that's due to low-volume really ;) there is alos a nice USB Audio kit! plus there is alittle board that is cheap and has two RJ45's on it already :) I'm myself studying the XC book at the moment. And geting familiar with the tool set :)~~ looks very exciting, cause these are the invovative chips! ok, may be an FPU is really missing on XCore, but how many DSPs have it anyway? well quite a few, but there was no FPU on dsps for ages! :)) also XC or C/C++ are so much more obvious then the bloody "menthal american military engeneers non-sense" called HDL-whatever! Cheers Everyone, Hope you will appreciate my excitment :)~ (l0l) -- ilya .d

15 years, 3 months

[LAD] Musescore "music trainer"?

by Camilo Polymeris

Hello all, I was a Smartmusic (http://www.smartmusic.com) user until that program stopped working with wine. For those of you who don't know it, it is a program that basically downloads finale files (score + accompaniment) from a centralized server for musical instrument practice. When you play along it uses either MIDI input or a microphone to rate your perfomance (which notes you played in tune, and so on). As far as I know, there is no native Linux or open source alternative. My questions are: Do you think it would be hard to code a plugin for musescore (http://www.musescore.org) to do the same pitch detection and comparison with the score? Or maybe would it be better to fork the code and make a stand-alone app, keeping it as compatible as posible? Any experience with pitch to midi? aubio (http://www.aubio.org) could be an option, but I haven't tried it. I think the music repository part could be addressed later... I don't have much time right now, but I'd like to ponder the posibility of working on it in the future. Maybe start experimenting in the meanwhile. Just an idea, Greetigs, Camilo

15 years, 3 months

Re: [LAD] panning thoughts

by Martin Leese

Eric Kampman <erickampman(a)me.com> wrote: > Hello, > > I'm writing a synth module on top of jack and I'm starting to contemplate > stereo. > > I looked up "pan law" and ... This reference: M.A. Gerzon, "Panpot Laws for Multispeaker Stereo", Preprint 3309 of the 92nd Audio Engineering Society Convention, Vienna (March 1992) looks at panning laws and psychoacoustics. The paper is concerned mostly with 3- and 4-speaker frontal stage stereo, but 2-speaker stereo is analysed as an introduction. Regards, Martin -- Martin J Leese E-mail: martin.leese stanfordalumni.org Web: http://members.tripod.com/martin_leese/

15 years, 3 months

Re: [LAD] What do you do for a living?

by Ralf Mardorf

-------- Forwarded Message -------- From: Ralf Mardorf To: Kris Calabio Subject: Re: [LAD] What do you do for a living? Date: Thu, 11 Nov 2010 16:23:46 +0100 On Wed, 2010-11-10 at 15:06 -0800, Kris Calabio wrote: > Hmm, lot's of anti Google sentiment here? I'm afraid that I was sort > of in the afterglow of positive testimonials when I wrote that > original email. What exactly does Google do that is unethical? Sure, > they're a huge corporation (bureucracy blah blah), but they are > companies that are a lot worse. > > It's great (and refreshing) to hear that a lot of us do what we do in > spite of money. community > capitalism > > Yeah, I'd like to > > work for Google, but who doesn't right? :) > > Not me... > > Gordon MM0YEQ > > > > > They cooperated with China, than there are trackers such as Google Analytics and read Fon's first reply. Off cause there are also some good sides of Google. - Ralf

15 years, 3 months

Re: [LAD] What do you do for a living?

by Kris Calabio

They're a corporation. Of course they're meant to make money. I know that already. Be more polite, please. ;) I'm afraid you're drew the wrong implication about myself from that statement, but perhaps I should have been more clear. I meant that Google likes to see open source in a resume. If they see that you developed for open source projects and talk about it in an interview (if you get one), they will be more likely to hire you. The Google representativs at the panel emphasized this point many times. I hope you understand where I was coming from now. Best, Kris --- On Wed, 10/11/10, fons(a)kokkinizita.net <fons(a)kokkinizita.net> wrote: From: fons(a)kokkinizita.net <fons(a)kokkinizita.net> Subject: Re: [LAD] What do you do for a living? To: "Kris C" <cpczk(a)yahoo.com> Cc: "Linux Audio Developers Mailing List" <linux-audio-dev(a)lists.linuxaudio.org> Date: Wednesday, 10 November, 2010, 2:44 PM On Tue, Nov 09, 2010 at 07:17:24PM -0800, Kris C wrote: > Google just loooooves open source and those involved. Quite naive. Google doesn't want to share its profits with Microsoft, that's all. For the the rest, it's and advertising machine meant to make money. The rest is just an means to this end. Wake up to reality, please. Ciao, -- FA There are three of them, and Alleline.

15 years, 3 months

[LAD] Pet-Project announce: AutoMate alpha1

by Harry Van Haaren

Hey guys, I've recently been working on a small MIDI automation line editor, and I've managed to get it into a working condition. Its a C++ & GTK jobbie, see this blog post for more info: http://harryhaaren.blogspot.com/2010/10/automate-another-stage-along-way.ht… I've not announced this as usable yet, as I am aware that there are some serious enough errors in it: 1. Drawing the GUI is not optimized and hogs resources. It always redraws the whole graph, and also redraws when its not necessary. 2. I'm not at all experience with threaded programming, and hence am pretty sure there's some threading possibilities to be explored. I've worked on this project on my own so far, and while it has taught me a lot, I think now I need to ask for some assistance with regards getting it polished for "everyday-use".. :-) If anybody has interest and want's to pull the repo to find out if it works for you, please provide feedback! (I'm 50% sure the waf script will need some tweaking, and maybe some other things too...) Cheers for reading, -Harry

15 years, 3 months

[LAD] Faster context switch

by Michael Ost

Hi list, I am seeing what appears to be a 4 - 7 usec context switch time on a 3 GHz Core 2 Duo machine with the 2.6 kernel, and 10 - 15 usecs on a 1.66 GHz Atom. Is that reasonable? Does anyone have any tips on how to speed that up, if possible? The background is that I have two apps --- one a straight Linux app and the other a Wine one. The Linux app is doing the mixing and the Wine app is hosting Windows VSTs. During audio processing I use a sem_t to signal from the mixer app to the vst app. Once the vst app is done processing, it signals back to the mixer app over another sem_t. I believe this is roughly the approach taken by other linux audio apps to host Wine vsts. And is also something like how Jack does IPC. But it takes about 4-7 usecs on the faster machine to signal another app and have it wake up again. I assume this is the time needed for a context switch. Since this has to happen twice per buffer, at a 32 sample buffer size and with 10 VSTs, overhead accounts for about 14% of the CPU. On the slower Atom machine, it would account for 33% of the CPU. Any hints on how to reduce that time, or an alternate IPC design to look in to? Thanks for any help, Michael Ost Muse Research, Inc.

15 years, 3 months

[LAD] Fwd: YTD-Meego on Googlecode - Planning Youttube direct uploading app

by Niels Mayer

Seeking Qt4.7 (esp. QtQuick/QML) and Linux multimedia experts to join a new project -- http://ytd-meego.googlecode.com -- to contribute code/planning/ideas for a port of http://ytd-android.googlecode.com/ to Meego running on http://en.wikipedia.org/wiki/Nokia_N900 and equivalent mobile computing platforms. FYI, the youtube direct application allows integrated capture/cataloging/uploading of video (or audio) to youtube captured directly from the handheld. See the following articles to understand how it is being used: http://gigaom.com/video/youtube-direct-abc7/ http://www.digitaltrends.com/computing/youtube-direct-is-helping-media-find… ---------- Forwarded message ---------- From: Niels Mayer <nielsmayer(a)gmail.com> Date: Sun, Nov 7, 2010 at 9:02 PM Subject: YTD-Meego on Googlecode - Planning Youttube direct uploading app To: qt-qml(a)trolltech.com I've created http://code.google.com/p/ytd-meego/ for the port of ytd-android to Meego using QtQuick/QML. YTD-Meego is to be a feature-equivalent port of the Youtube-direct application for Android ( http://ytd-android.googlecode.com/ ). Initially, the purpose of the googlecode project will be for issue-tracking and planning features and their implementation in QtQuick. The subversion repository will eventually contain working snippets of QtQuick test code that will evolve into an application with the help of the community. Please send mail if you want to be added as a project contributor/committer for contributing ideas, code snippets and helping developing this application openly. -- Niels http://nielsmayer.com

15 years, 3 months

[LAD] [ANN][Request] AlsaPlayer-0.99.81 is out !

by Dominique Michel

I send the following to the LAA : This is a bug fix release. A big thanks to all that send fixes to me, without you, the AlsaPlayer will not be living. See the ChangeLog for the details and the names of the contributors : http://alsaplayer.svn.sourceforge.net/viewvc/alsaplayer/trunk/alsaplayer/Ch… I will also stress you to contribute to the AlsaPlayer. At least 2 things need to be fixed. The first one is the jack output plugin that is using deprecated functions. The second one is the resampling. For that, libsamplerate will give a better quality. Enjoy the AlsaPlayer, # # # # # To quote Fons (he do have a better English than mine) : While it's a nice player, it has some serious audio quality issues. - Resampling 44.1 -> 48 kHz (for jack) sounds horrible... - The sndfile input plugin reduces everything to 16 bits. This is really absurd, even if your files and your sound card are 24 bit you only get 16. Floating point wav files apparently aren't read at all (they load but produce silence when played). All of this could be solved by using a good resampler lib, and making the internal format floating point rather than short. # # # # # This change can be made using libsample rate. This is a must have feature for me, and it must be implemented before any other change because it will ease further development (it is a lot of free re-usable audio code in float). I am just an admin and don't have the knowledge to make the needed code. Anybody that can contribute such a great feature for AP will be welcomed. If you are interested, please take contact with me, on this list or privately. Ciao, Dominique Michel -- "We have the heroes we deserve."

15 years, 3 months

[LAD] PCM Troubleshooting Questions

by Rory Filer

Hello, For most of the last week I've been stuck trying to configure my silicon peripheral to generate Normal I2S audio data to mostly _bad_, results and had a couple of questions (below). Briefly, my setup is as follows: SqueezeServer on a Windows host PC, Squeezeslave on my embedded Linux device sending audio samples through the kernel (2.6.28) down to my driver which lives in sound/soc/pxa.I've been trying to configure my device for I2S format. Outside the CPU, the I2S-formatted PCM data is sent to an external FM transmitter chip so I should be able to hear the audio on my FM receiver. Since the only "new" stuff here is my driver and the output hardware, I've been focussing my attention on them until now, but haven't had much success hearing anything. I've actually heard _some_ recognizable audio - I could tell it was the song I had selected, but it sounded very distorted (sounded horrible) and I'm certain it wasn't because of a weak FM transmitter signal. I'm guessing it was a mismatch in the formatting of the I2S data between what I was sending and what the FM Tx chip was expecting and that has formed the basis of my troubleshooting effort. I've run out of ideas for things to try at the bottom end and now I want to make sure my top-end components are working properly. This is weird because I've stopped being able to hear anything lately, but a scope confirms my I2S lines are all lit up. Here are a couple of questions... 1) The CPU supports a packed mode write to the Synchronous Serial Port (SSP) FIFO, meaning that two 16 bit samples (one left channel and one right) can be written at the same time, both being packed into a single 32 bit FIFO write. My driver enables this mode, but my question is, where in the kernel would the samples be combined into a 32 bit chunk for writing? I'm using Squeezeslave on my embedded device as the player and I've checked the sources and it doesn't look like it's happening in there. Makes more sense to be somewhere further down in the stack so players don't have to care about the details of the hardware they are running on. I was wondering if anyone knew where this packing might take place. 2) Are their any useful tools out there for debugging/narrowing down where problems in the audio path might lie? My player is an embedded platform and I've only ported Squeezeslave to it, but for all I know there could be a problem anywhere from SqueezeServer, through Squeezeslave, down into the stack, my PCM driver or even the FM transmitter. To eliminate the latter as a problem I'm looking for another device that spits out known good I2S audio and I'll feed that into the FM Tx and hopefully eliminate that. But there's a lot of code from the SSP back and it would be great if I had some simple tone generator application (for instance) that was easily portable to an ARM9 platform (kernel 2.6.28) that I knew was sending correct data down the stack. 3) My experience with Linux and audio is just beginning and so far it's been right down at the driver level, so a question about audio playing software: when a player produces a PCM stream from, say, an MP3 file, does it automatically interleave the left channel and right channel or does it produce two separate streams, one for left and one for right? I can't tell from reading the Squeezeslave code, but it looks like the audio data is sent in one continuous stream, so ... interleaved? 4) For those of you experienced with I2S and other PCM formats, what would a Normal I2S stream sound like on a DAC that thought it was receiving Justified I2S? Would the audio still be intelligible or would you hear nothing at all? Thanks to all who read this post. Cheers, Rory

15 years, 3 months

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Linux-audio-dev