Linux-audio-dev

linux-audio-dev@lists.linuxaudio.org

7516 discussions

[linux-audio-dev] How does one alter the main mixer volume (or any other mixer level) in ALSA from a command line (via a simple system call)?

by Ivica Bukvic

Hi all, I have this small question: Is it possible (I presume it is) to alter mixer settings in Alsa by invoking some kind of a system call using shell (i.e. how the RME hdsp can have its stuff altered)? If so, what is the range of values that describe the loudest and softest levels? Finally, are these levels standardized? Any help on this matter is greatly appreciated! Thank you! Sincerely, Ico P.S. Apologies for cross-posting.

22 years, 8 months

Re: [linux-audio-dev] question re: hammerfall cards

by Patrick Shirkey

>its completely ALSA-generic. it applies to all cards. the >documentation that exists right now is weak. the "best" there is so >far (that i've seen) is in alsa-lib/doc/asoundrc.txt. its not for the >faint-of-heart. its not even for people who cycle double centuries for >fun :) If you want to know more about the .asoundrc file then have a look at http://www.alsa-project.org/alsa-doc/doc-php/asoundrc.php3?company=Generic&… This doesn't explain aplay/arecord in full but there are some examples. -- Patrick Shirkey - Boost Hardware Ltd. Http://www.boosthardware.com Http://www.djcj.org - The Linux Audio Users guide ======================================== Being on stage with the band in front of crowds shouting, "Get off! No! We want normal music!", I think that was more like acting than anything I've ever done. Goldie, 8 Nov, 2002 The Scotsman

22 years, 8 months

[linux-audio-dev] Jack 0.61.0 release

by Taybin

Jack Audio Connection Kit 0.61.0 Release The Jack team is pleased to announce the release of version 0.61.0 of the Jack low-latency audio server. Jack allows applications to share data and audio devices in synchronous operation, and has already seen a year of hard testing and refinement. The API has stabilized for the foreseeable future, although backwards compatibility is not guaranteed. More information on Jack is available at the group's web site, http://jackit.sourceforge.net/. What's new: * In-process client support. * documentation updates. * bug fixes. Developers and users interested in Jack should sign up to jackit-devel, our mailing list.

22 years, 8 months

[linux-audio-dev] [ANN] ladcca 0.3.1

by Bob Ham

Just some build fixes. Should have released this before but it slipped my mind. * fixed problem with the readline headers paying attention to HAVE_CONFIG_H * fixed HAVE_GTK2 always being defined to work properly and only be defined if it's available http://pkl.net/~node/ladcca.html Bob -- Bob Ham <rah(a)bash.sh>

22 years, 8 months

[linux-audio-dev] [OT] [for sale] RME Hammerfall (digi9652)

by Paul Davis

(sorry for the off-topic post, but i want people on these lists to have the first chance at this) Rev 1.5 Hammerfall Digi9652 all cables Windows/Mac driver CD original packing materials current street price: about $580 asking price: $480 OBO + shipping

22 years, 8 months

Re: [linux-audio-dev] Fwd: CSL Motivation (fwd)

by Stefan Westerfeld

Hi! On Thu, Feb 27, 2003 at 12:39:29PM -0500, Paul Davis wrote: > >No. PortAudio makes a lot of choices for the software developer, and thus > >provides an easy abstraction. > > The point is that PortAudio follows the same basic abstraction that > the audio APIs on the overwhelmingly dominant platforms for audio > software development follow (callback-driven). Those APIs have emerged > from years of practical experience writing serious audio applications, > and I strongly believe (as you know) that they should be taken very > seriously. The fact that a couple of early linux hackers felt that the > open/read/write/close/ioctl model was the right thing seems to me of > much less significance than the number of working, useful, > sophisticated software that have been built around a set of > abstractions that are more or less identical to the ones PortAudio > provides. I take callback APIs very seriously, because they provide an excellent base for developing music applications. In fact, artsd is entierly based around the concept of callback APIs (you have calculateBlock functions with float data blocks which are called in a graph of connected synthesis modules). I do have years of practical experience writing such software ; and I'd say Tim Janik is not worse than me on this front (www.arts-project.org and beast.gtk.org). Of course, thats together only 16 man-years of learning audio development, not full-time, even. However, while discussing our experience on the list might give people an idea whether or not they should even bother listening to our opinion. Ultimately, I would however prefer to discuss the technical aspects of CSL ;). The open/read/write/close/ioctl as you say it is just an _interface_ between the kernel and the application, because on linux, we do things this way, rather than having any other means of doing callbacks from kernel space to user space. I do agree that it makes a _poor_ API for most low-latency realtime music software, because its extremely hard to use. However, you must also admit that it _is_ capable of mapping the features that the hardware has to the software without additional buffering. If you combine it with memory mapping (like linux supports), you can write to the soundcards buffer directly, and get the hardware interrupts the card generates via ioctl. Quake for instance uses this to prerender a lot of audio and rerender it once a new sound needs to be mixed into the soundstream. It thereby can combine robustness (which is needed on a timesharing system such as unix) with extremely high responsiveness. This is not something you can do with an easy callback API. All that said, note that CSL _has_ a callback API in addition to the normal one. Its just that CSL is somewhat more low level than PortAudio in the sense that it allows you to access the open/read/write/close/ioctl stuff more directly, which has two effects (a) you can get all low level features (such as fragments) linux provides for you and deal with them yourself if you want to (b) you can implement the callback API you need ontop of it if you want to ; that is: you can implement PortAudio on top of CSL without any loss of performance (through extra copies and such) - you can however _not_ implement CSL on top of PortAudio without any loss of performance > > This will mean, however, that actually porting > >software to PortAudio will probably be hard (compared to CSL), > > that depends on where you started from. if you started from ASIO, it > won't be hard. if your app started as a VSTi, it won't. but sure, if > you started as your typical linux audio app that reads a stereo file > from the disk and shoves it into the audio interface, it will be > fairly easy to use CSL and a bit harder to use PortAudio. > > this porting thing bothers me. we have only a handful of really great > apps for audio under linux right now (though we have lots of rather > interesting ones) and most of them already can use JACK. i would much > rather see people forced (just like apple have done) to work with the > "correct" abstraction than continue with multiple wrappers and > multiple different abstractions as we move into a period where i hope > to see an explosion in linux audio applications. You can not force people to do the right thing in the free software community. They will continue doing whatever they feel like doing. And I like that aspect of free software development. If you provide something that is convenient for them to use, they will use it. Thus, I'd rather like to provide them with CSL, instead of having them use OSS all the time ; I hate the OSS API, and the fact that 99% of all free software supports the OSS API seems to be an indication to me that if you provide something "like" the OSS API (and CSL can do this), then 99% of all free software developers at least would be willing to also support it, if it provides no additional overhead over the OSS API ; or maybe even switch. > > whereas > >writing new software for PortAudio might be convenient, _if_ the software > >falls in the scope of what the abstraction was made for. > > well, there are at least two sets of evidence to consider there. i > think there is plenty of evidence in the world of windows that a large > amount of interesting audio software works with the ASIO and DirectX > models (which are semantically similar, if syntactically worlds > apart). its not limited to audio synthesis. but at the same time, its > worth noting that "most" apps on windows that emit audio for some > reason (i.e the ones that are not actually audio applications) do not > use ASIO or DirectX. so there is evidence to support the idea that at > least a couple of abstractions are necessary. > > in contrast, CoreAudio offers only 1 abstraction model as i understand > it. so apple at least appear to have been willing to bet that all > software can use a single model. I think the point is similar to that of programming languages: all programming languages that are turing complete are equivalent in the sense that you can implement everything in them. Thus, you can choose to write KDE or Ardour in Scheme, C++, C, Assembler or Java. However, there are differences in convenience and performance for a given task ; thus if you have extreme performance requirements, you will need to take some inconvenience on you, if you don't, you can choose whatever you like. When it comes to audio abstractions, music applications (the low-latency kind-of thing) do have extreme performance requirements ; thus you want to have immediate notification on the buffer state, which you can do by a callback API (under linux/unix, this pretty much means an extra, high priority thread using select/write with careful fragment size/count settings). Other applications can use whatever is convenient for them ; thus some use this and some that. As long as CSL can provide what is required to support all applications in an efficient manner, I see no reason why we should force everybody to use just one abstraction (we can't anyway). And even if CoreAudio is only 1 model, around which all other apps are based, I would say that there will be wrappers, which also provide new abstractions on top of it. Even PortAudio has a layer (pablio) to emulate blocking read/write, probably because somebody felt it was needed. > >[... JACK ...] > > i don't think its about liking it so much as the fact that it does > some things that are extremely important to many of us and that > nothing else can do. Yes, I think there are few people who like something that is not useful to them. > > If some people will like CSL, why not? > > well, i am *really* not trying to be argumentative just for the sake > of it, but ... the reason not to is that CSL doesn't offer any new > functionality. its just another wrapper, and we already at least one > of them, with some evidence that its capable of supporting all apps. PortAudio can support all applications at extra performance costs, because emulating a blocking API on PortAudio does require extra buffering. Especially for low latency stuff, this is problematic. For instance, I would say that artsd could not use PortAudio without extra latency costs, because it needs to integrate its network-aware mainloop with the audio mainloop. I doubt thats possible with PortAudio, whereas aRts does already support CSL. > >On the other hand, if you added JACK support to CSL, you could also mix the > >output of all of these "sound servers" into JACK, without endangering your > >latency properties. > > since JACK has no desire to be a general purpose API for all apps, > this would be more appropriate than the other way around, i > think. there is no reason to run a JACK system that does audio i/o via > a sound server - it just isn't going to work with the performance > promises that JACK is supposed to provide. Well, you might want to run a professional audio APP with suboptimal latency if you're a home user, and just interested in composing a song. You don't need extremely low latency for that, but you might want to use a "professional APP". But I agree with you. A JACK backend for CSL is higher priority than a CSL backend for JACK. > i think that the fundamental problem here is the division between: > > * the kinds of apps that emerge from the repeated questions on gtk-list > (and i'm sure the KDE equivalent): "what's the best way to play a > sound?" > > * "serious" audio applications (and, i suppose, video apps too) > > CSL seems to be about providing something to the first group that > PortAudio *could* provide, even though it may not be the best or > easiest method (although it might be the most appropriate). but i > really don't see how it offers anything to the second group that want > features (and to some extent, *lack* of features) beyond those CSL or > even PortAudio provide. > > i see it this way because CSL doesn't appear to me to have emerged > from efforts of audio developers. its come mostly from the desktop > world as way to meet the need for a "cross-desktop" way to say "you've > got mail" or play the KDE startup theme, and other "singular" audio > events. As I said, I consider myself an audio developer. But regardless of what I consider myself or what you consider me, I can't improve CSL but on techincal facts. Its irrelevant where it came from or who wrote it. I just want to make it perfect for the purpose it has. If you'd say it isn't right now, I would be more than happy if you let me know that needs to be improved. > as i said above, the windows experience suggests that it may be > useful, perhaps even necessary, to have two different abstractions to > support these two rather different classes of applications. but the > jury is out on that. Well, its important to clarify that we need to distinguish two things: (a) the interface the operating system provides (b) the interface the application is using For the scope of CSL, (b) is not too important. We only want to wrap the operating system interface with little overhead, so that you can provide any abstraction the application will need on top of that, if its different than what CSL provides. The operating system (linux) currently provides these: - a callback API (select/write) - a blocking API (write) - a non-blocking non-copying non-linear callback-aware API (mmap) Depending on how you handle concurrency in your application (event-based single threaded, not at all, multi threaded), and what your audio needs are, any of these might be optimal for you. CSL is realizing this, and currently wraps the first two ; there already have been some discussions on whether we also need to provide a pwrite like write call in CSL to handle the third abstraction as well. Cu... Stefan -- -* Stefan Westerfeld, stefan(a)space.twc.de (PGP!), Hamburg/Germany KDE Developer, project infos at http://space.twc.de/~stefan/kde *-

22 years, 8 months

[linux-audio-dev] question re: hammerfall cards

by Thomas Vander Stichele

Hey, I just got a new box and two hammerfall 9636 cards for a project at work. The motherboard is a KT4 Ultra that has some onboard audio chip I don't care about but can't turn off either in the bios. I've put in the first hammerfall card, got alsa 0.9.0 rc7 from the freshrpms site (it's a redhat box), and started configuring things. I've got the feeling I'm missing some things, so here are some simple questions. a) It looks like the Hammerfall driver doesn't have a mixer interface, is this correct ? Here's an ls in the relevant dir: [root@framboos asound]# ls card0/ id pcm0c pcm0p rme9652 b) It looks like the onboard audio chip is controlled by an OSS driver, it doesn't show up in the alsa drivers either, which is fine by me, since I'm not going to use it. Is there any problem with OSS modules being loaded at the same time as ALSA modules ? c) I bought the card so I could record optical S/PDIF. The manual says I need to tell the card that I want the ADAT1 Input source to be Optical S/PDIF. Right now I get, in /proc/asound/card0/rme9652: [root@framboos card0]# cat rme9652 RME Digi9636 (Rev 1.5) (Card #1) Buffers: capture cdc00000 playback cda00000 IRQ: 5 Registers bus: 0xde000000 VM: 0xd088a000 Control register: 44008 Latency: 1024 samples (2 periods of 4096 bytes) Hardware pointer (frames): 0 Passthru: no Clock mode: autosync Pref. sync source: ADAT1 ADAT1 Input source: ADAT1 optical IEC958 input: Coaxial IEC958 output: Coaxial only IEC958 quality: Consumer IEC958 emphasis: off IEC958 Dolby: off IEC958 sample rate: error flag set ADAT Sample rate: 44100Hz ADAT1: No Lock ADAT2: No Lock ADAT3: No Lock Timecode signal: no Punch Status: 1: off 2: off 3: off 4: off 5: off 6: off 7: off 8: off 9: off 10: off 11: off 12: off 13: off 14: off 15: off 16: off 17: off 18: off Which I read as the Input source being ADAT optical instead of S/PDIF. How do I set it to S/PDIF ? d) the card came with a sub-D-connector that connects to the card's 15 pin port, which branches off two RCA jacks. I don't suppose these RCA jacks provide an analogue output by any chance on which I can monitor for sound ? I'm sure I'll bug you with more questions later on, but these are the most pressing ones at this point. Any help is greatly appreciated. Thomas -- The Dave/Dina Project : future TV today ! - http://davedina.apestaart.org/ <-*- thomas (dot) apestaart (dot) org -*-> Oh, baby, give me one more chance <-*- thomas (at) apestaart (dot) org -*-> URGent, the best radio on the Internet - 24/7 ! - http://urgent.rug.ac.be/

22 years, 8 months

[linux-audio-dev] LADSPA hints (part II)

by Paul Davis

i like the recently suggested new hints. i'd like to add the following, which would allow hosts to do the right thing with both SooperLooper and SC4, which are the first (only?) LADSPA plugins to have control output ports, but use them in very different ways. its almost impossible as it stands to build a GUI for them both that does them justice. --p /* Hint LADSPA_HINT_OUTPUT_METER indicates that the value of output control port is likely to be most meaningful to the user if displayed as a meter. Can be combined with LADSPA_HINT_LOGARITHMIC if the meter should use a log scale (e.g. dB). */ #define LADSPA_HINT_OUTPUT_METER 0x?? /* Hint LADSPA_HINT_OUTPUT_VALUE indicates that the value of output control port is likely to be most meaningful to the user if displayed as some kind of numeric value display. */ #define LADSPA_HINT_OUTPUT_VALUE 0x??

22 years, 8 months

[linux-audio-dev] Update On PDAudio (Portable Digital Recorder)

by Len Moskowitz

We've posted an update about our forthcoming 24/192 portable recorder: http://www.core-sound.com/HighResRecorderNews.html It runs on an iPAQ and the Linux version of the software uses Familiar/GPE and ALSA. Thanks to all for the high level of interest! Len Moskowitz Core Sound

22 years, 8 months

[linux-audio-dev] POSIX clocks now in linux 2.5

by Bob Ham

>From the 2.5.63 changelog: o POSIX clocks & timers Phat :) -- Bob Ham <rah(a)bash.sh>

22 years, 8 months

Jump to page:

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

Linux-audio-dev