Linux-audio-dev

linux-audio-dev@lists.linuxaudio.org

7516 discussions

by Richard Spindler

Hi, on my quest to understand the ALSA API, I created a little Image to get an overview about the various states and how they relate to each other. http://propirate.net/oracle/bilder/ALSA.png Hopefully I got everyting right, can you have a look at this image and tell me whether this is correct? Thanks -Richard

19 years, 7 months

[linux-audio-dev] select() ?? before snd_pcm_writei(handle, buffer, frames);

by kurmisk

Hi Developers. I have write my small _alsa_test_program_. [C code ckunkz see below] It works good but now i wanna before call rc = snd_pcm_writei(handle, buffer, frames); somehow check - is sound device free for this call or not. For OSS i have such problem solve with select() : fdo = open("/dev/dsp", O_WRONLY ); ..... FD_ZERO(&wfds); FD_SET( fdo , &wfds); Zeit.tv_sec=0; Zeit.tv_usec=1; response=select(fdo+1, NULL, &wfds, NULL, & Zeit ); Tnx in advance Alf int alsa_snd_pcm_open_write(unsigned int sample_rate__, unsigned int chan__ , snd_pcm_uframes_t * frames_p , snd_pcm_t ** handle) { int dir; // snd_pcm_t *handle; snd_pcm_hw_params_t *params; unsigned int val; int ret_fun; int chan_nr; snd_pcm_uframes_t frames; /* Open PCM device for playback. */ ret_fun = snd_pcm_open(* &handle, "default", SND_PCM_STREAM_PLAYBACK, 0); if (ret_fun < 0){ fprintf(stderr, "unable to open pcm device: %s\n", snd_strerror(ret_fun) ); exit(1); } /* Allocate a hardware parameters object. */ snd_pcm_hw_params_alloca(&params); /* Fill it in with default values. */ snd_pcm_hw_params_any(* handle, params); /* Set the desired hardware parameters. */ snd_pcm_hw_params_set_access(* handle, params, SND_PCM_ACCESS_RW_INTERLEAVED); /* Interleaved mode */ snd_pcm_hw_params_set_format(* handle, params, SND_PCM_FORMAT_S16_LE); /* Signed 16-bit little-endian format */ //snd_pcm_hw_params_set_format(handle, params, SND_PCM_FORMAT_S16_BE); chan_nr = chan__; snd_pcm_hw_params_set_channels(* handle, params, chan_nr); val = sample_rate__; snd_pcm_hw_params_set_rate_near(* handle, params, &val, &dir ); frames = * frames_p; //frames = 2048; /* Set period size to __nr__ frames. */ snd_pcm_hw_params_set_period_size_near(* handle, params, &frames, &dir); * frames_p = frames; /* Write the parameters to the driver */ ret_fun = snd_pcm_hw_params(* handle, params); if (ret_fun < 0) { fprintf(stderr, "unable to set hw parameters: %s\n", snd_strerror(ret_fun)); exit(1); } /* Use a buffer large enough to hold one period */ snd_pcm_hw_params_get_period_size(params, &frames, &dir); snd_pcm_hw_params_get_period_time(params, &val, &dir); } int main(int argc, char **argv) { ........ snd_pcm_uframes_t frames; unsigned int sample_rate; unsigned int chan; snd_pcm_t *handle; //---- sample_rate=44100; chan=2; frames=1023; // 4096 // frames=2047; // 8192 // frames=4095; // 16384 // frames=8191; // 32768 alsa_snd_pcm_open_write( sample_rate , chan , & frames , & handle); ....... while (1) { ....... rc = snd_pcm_writei(handle, buffer, frames); if (rc == -EPIPE) { /* EPIPE means underrun */ fprintf(stderr, "underrun occurred\n"); snd_pcm_prepare(handle); } else if (rc < 0) { fprintf(stderr, "error from writei: %s\n", snd_strerror(rc)); } else if (rc != (int)frames) { fprintf(stderr,"short write, write %d frames\n", rc); } ....... } snd_pcm_drain(handle); snd_pcm_close(handle); }// main ------

19 years, 7 months

[linux-audio-dev] Alsa Problem

by Richard Spindler

Hi, I'm still trying to grok alsa, and now I have the following problem: I'm using snd_pcm_writei to playback some audio, however, after a little time the call fails with the error: "File descriptor in bad state", which I believe I cannot recover from. Why does this happen, and what could I do about this? cat /proc/asound/card0/pcm0p/sub0/* prints the following stuff, I don't know whether this is helpful? access: MMAP_INTERLEAVED format: S16_LE subformat: STD channels: 2 rate: 48000 (48000/1) period_size: 8192 buffer_size: 16384 tick_time: 1000 card: 0 device: 0 subdevice: 0 stream: PLAYBACK id: VIA 8235 name: VIA 8235 subname: subdevice #0 class: 0 subclass: 0 subdevices_count: 4 subdevices_avail: 3 64 state: RUNNING trigger_time: 1143205215.076663000 tstamp : 1143205216.323831000 delay : 13856 avail : 2528 avail_max : 14368 ----- hw_ptr : 59872 appl_ptr : 73728 tstamp_mode: NONE period_step: 1 sleep_min: 0 avail_min: 8192 xfer_align: 8192 start_threshold: 1 stop_threshold: 16384 silence_threshold: 0 silence_size: 0 boundary: 1073741824

19 years, 7 months

[linux-audio-dev] Re: [linux-audio-user] [ANN] Kontroll

by Florian Schmidt

On Mon, 27 Mar 2006 00:49:04 +0200 Florian Schmidt <mista.tapas(a)gmx.net> wrote: > http://tapas.affenbande.org/?page_id=42 Oh and i forgot to ask a question: What is the canonical way for a gtk app to receive hotkey press events regardless of window focus? Flo -- Palimm Palimm! http://tapas.affenbande.org

19 years, 7 months

[linux-audio-dev] [ann] PDX7 V2 - A 6-Op-FM synthesizer for Pd

by Frank Barknecht

PDX7 V2 In 2001, I started learning Pd. The first bigger patch I did was a six operator FM synthesizer loosely modelled after the famous DX7, which I called PDX7. Now it's five years later and not only in retrospect this first patch is quite a mess. But I still like the sounds it generates. In the meantime I learned a lot more about Pd and about a lot of other things. So I thought, why not celebrate the fifth anniversary of me doing Pd, the fifth "birthday" with a cleaner version of the PDX7, that maybe others can understand to use as well. The result is the PDX7 V2. Read and hear more at: http://footils.org/cms/show/51 Ciao -- Frank Barknecht _ ______footils.org_ __goto10.org__

19 years, 7 months

[linux-audio-dev] [ANN] Snd-ls V0.9.5.4 and Das_Watchdog V0.2.1

by Kjetil S. Matheussen

Download from http://ccrma.stanford.edu/~kjetil/src/ Snd-ls v0.9.5.4 ================ Contains -------- Snd v7.15 from 17.8.2005 About ----- Snd-ls is a distribution of the sound editor Snd. Its target is people that don't know scheme very well, and don't want to spend too much time configuring Snd. It can also serve as a quick introduction to Snd and how it can be set up. Changes 0.9.5.3 -> 0.9.5.4 -------------------------- -Changed default resampling quality to SRC_SINC_BEST_QUALITY -Added workaround for shift-handling across various keyboard settings. (shortcuts for zoom in and undo works with american keyboards now.) -Added check for Guile 1.8. Snd-ls crashes with guile 1.8. (all versions I have tried of 1.7 seems to work though...) -Use JackPortIsPhysical instead of "alsa_pcm" when finding jack ports. -Updated the rt stuff to latest versions. *************************************** *************************************** Das_Watchdog V0.2.1 =================== About ----- Das_Watchdog is a general watchdog for the linux operating system that should be runned in the background at all times to ensure a realtime process won't hang the machine. Changes 0.2.0->0.2.1 -------------------- *Cleaned up source a bit. *Properly find number of timer processes. *Added shortcuts for optargs.

19 years, 7 months

[linux-audio-dev] [ANN] netjack-0.10

by torbenh＠gmx.de

netjack-0.10 comes with a huge quality improvement to alsa_in and alsa_out. its now ready for prime time. try alsa_out -f 10000 and listen... -- torben Hohn http://galan.sourceforge.net -- The graphical Audio language

19 years, 7 months

[linux-audio-dev] python and jack, performance report

by Leonard "paniq" Ritter

-- oops, the first one was sent as a reply, here's the whole thing as a new thread. frustrated by the poor implementation of the jack bindings for python (pyjack), i wrote my own in native python using ctypes. the first test client mixed a 440hz sine wave using native python lists, and the cpu usage was about ~11%. i reimplemented the sine generator with numerics, and got it down to ~2%. i believe that considering the overhead of the python implementation, that result isn't too bad, and maybe allows for more than just prototyping. i attached the jack wrapper with the test client contained for those who are interested. its not entirely wrapped and lacks some functionality. -- -- leonard "paniq" ritter -- http://www.mjoo.org -- http://www.paniq.org

19 years, 7 months

[linux-audio-dev] Linux Audio Conference 2006: Register now!

by Frank Neumann

Hi all, this mail is just to inform you that registration for the 4th International Linux Audio Conference (or "LAC2006" for short), APril 27th-30th, in Karlsruhe, Germany, is possible as of now by visiting our web page at http://lac.zkm.de and clicking on the link "Registration" on the left. Like last year (and the years before), registration is free and is only necessary because we need to prepare nametags, make estimations about room sizes/required chairs and want to learn a bit about our audience. That's all, honestly! :-). Information given there will NOT be used for any kind of selling, spamming or otherwise abusing. Also, the preliminary conference program has been published a few weeks ago, available through the same link give above. Thanks for registering in case you plan to come! Greetings, Frank Neumann Goetz Dipper LAC2006 organization team

19 years, 7 months

Re: [linux-audio-dev] Best opensource license to use?

by Chris Cannam

> i intend to release an application, which allows to > use VST(i)s running on an XP machine from > a linux jack client, hooked up over ethernet. If all the code is your own, consider the approach we used in dssi-vst, which is to use the GPL plus an exception to allow for the unavailability of VSTSDK sources. It won't be GPL-compatible or "Debian" free, but nothing will be if not all the source is available, and I think it best expresses the intention -- if your preference would naturally be for the GPL, that is. Meanwhile, write and remind Steinberg that their license could be more helpful to developers of Windows and Linux plugins and hosts. It won't be news and there's no real reason they should care, but it's worth a nudge that these things matter to people, including those who would like to promote, recommend and use their standard. As it happens I'm also shortly about to release a Windows plugin host that is not at all in competition with Steinberg products, but that can't host VSTs for licensing reasons and will use DSSI plugins instead. Hosting audio plugins is not a major part of this program so the difference isn't going to be all that significant to me or anyone else, but it's an interesting situation. Chris

19 years, 7 months

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Linux-audio-dev