Linux-audio-user September 2009

linux-audio-user@lists.linuxaudio.org

143 participants
148 discussions

by Scott

I'd really like to use Audacity through Jack. Unfortunately it's behavior complicates this. I can load up a file, but nothing shows up in jack connections until I hit play. Then two PortAudio channels appear! If I stop playback they disappear. The next time I press play they appear again, but incremented by 2 and, of course, not connected to my out channels. This was all covered in an old thread here: http://linuxmusicians.com/viewtopic.php?f=28&t=552 The best solution I could find there was to use PatchBay. That kind of works, except the two PortAudio output sockets created by Audacity automatically connect to the first two input sockets on my system which don't happen to be where the monitor cables plug in (they're in 10 and 11). Is there a way to specify which output plug routes to which input plug via patchbay or is it stuck at the socket level with no finer-grained control? In a pinch, is there another mp3 capable audio player that works with jack without jumping through any more hoops than just starting the app? -Scott p.s. Using PortAudio v19, Audacity 1.3.5-beta

16 years

jack command line and xml patchbay files

by Grammostola Rosea

Hi, I isn't possible to start a patchbay xml file, made in qjackctl, with jack using the command line. Is such a thing totally unthinkable? If not, wouldn't it be a nice feature? Is there another way to make connections with jack using the command line? Regards, \r

16 years

Sound newbie question re: PCM volume control ??

by William Case

Hi; I have been lurking on this list for a month or so hoping to learn something about sound drivers, alsa, etc. But I do have a question that is mystifying me. Why is there a volume control for PCM? Isn't PCM an on/off thing? Either you need to sample a stream or you don't? Shouldn't alsa code just use PCM as needed without user intervention? Is there real instances where users need easy access to PCM settings? All my playback devices have their own volume control; usually two or three controls i.e. software volume control in alsamixer, an application volume control and often a physical control on the hardware. Computer generated sound is complex enough without throwing in unexpected extra volume controls. A secondary question for experienced sound users: How, or in what order or priority, do you usually the adjust volume levels of alsa, application and physical (knob or button on speakers etc.) controls? So far I just have been twiddleing a little of this or more of that with no real rationale. -- Regards Bill Fedora 11, Gnome 2.26.3 Evo.2.26.3, Emacs 23.1.1

16 years

Control GUI-less VSTi?

by Simon Wiesmann

Hello list members, I'd like to use the Magical 8bit Plug VSTi from YMCK, but it doesn't have its own GUI. http://www.ymck.net/english/download/index.html Is there a VSTi host that allows controlling the VSTi parameters without a GUI? I know there's other similar lo-fi VSTi, but most of them are too complicated for me as I have no knowledge whatsoever of synths. Thanks a lot! Simon

16 years

Denemo notation and Midi

by TheOther

Nils pointed out the Denemo was good for Midi notation. I've been using Rosegarden 1.7.3 under Fedora 9 and Lilypond for printing with excellent success. When I upgraded to Fedora 10, printing from Rosegarden to Lilypond generated a Lilypond error, the dialog box that was to say what went wrong in Lilypond was empty. I don't what went wrong. As far as I know, the versions of Rosegarden and Lilypond didn't change. So I looked into Denemo. The first difficulty was trying to import the Midi file I exported from Rosegarden. I couldn't find an import function under the File menu. I went through File, Edit, View, Mode, Input (under Input-> Midi Input, selected that and got a Pitch Input Control box which did nothing to Import a Midi file), Playback, More, and Help (tried the Browse Manual F1 selection and got the error "Could not find Mozilla in the path" -- puzzling, I'm using Firefox 3.0.13) and that was all the options at the top. I then mouse'd over the next line with the icons and found something that looks like an input tray icon. Tried that and got the Open a File for input. Okay, a little bit hidden there trying to input a Midi file, but at least I'd found the function I was looking for. In Rosegarden I'd exported my Midi (with Lyrics) file in .mid, .ly, and .xml. Denemo crashed in trying to open the .mid file, the .xml files Denemo use are a special version adjusted to Denemo only (.denemo, and .dnm) and I had success in opening the .ly Lilypond file. Unfortunately, Denemo didn't show the Lyrics when it loaded the .ly file. Does Denemo work with Lyrics at all? Otherwise, may I suggest an added feature? Best, Stephen.

16 years

midi over wine

by Patrick Shirkey

Hi, Has anyone had any success connecting wine apps with midi output to native apps on Linux? Cheers. -- Patrick Shirkey Boost Hardware Ltd

16 years

Re: [LAU] Denemo notation and Midi

by list＠nilsgey.de

Am 08.09.2009 um Uhr haben Sie geschrieben: > Nils pointed out the Denemo was good for Midi notation. Thanks for your interest! But to be accurate, I said: "If you need decent Midi: Use Denemo in 2 months. " > tried the > Browse Manual F1 selection and got the error "Could not find Mozilla > in the path" -- puzzling, I'm using Firefox 3.0.13) and that was all > the options at the top. The standard Mozilla/Firefox binary is "firefox". For some reasons we still have "mozilla" as binary name in Denemo. You can change your browser in Edit -> Preferences -> Externals You are right that its not obvious how to load non-denemo files. According to our roadmap a gui overhaul is planned for 0.9, which is the version after the current one. Your problems will be remembered. > Does Denemo work with Lyrics at all? Otherwise, may I suggest an added > feature? Denemo has a good lyric editor with multiple stancas per staff and it shows you where you syllabes will be printed. In Denemo only the lyrics for the current selected staff will be shown. Its in the Object-Menu (Denemos main menu-bar where you find all the functions) in Lyrics -> Add Lyrics Now to the important part: Importing Midi and Lilypond is not as good as it should be, but currently its more important for us to get things out of Denemo than get things in. We know that many users want to convert their old files at first, but we do not have enough manpower to work on everything at once, sorry. Please be patient. So while its possible to get midi and lilypond from external files it can be quite some work for the user to overcome some crashes and splitting the source file in "easier" packages. Additionally you can copy and paste from a lilypond file directly to Denemos staffs. Please use Edit -> Paste lilypond notes for that. I have not tested that personally so I don't know what can be copied with that and what not. If you have any questions or wishes, please tell us. For me its ok to answer Denemo questions here, but we also have our own mailing-list in Denemo. Nils http://www.denemo.org

16 years

Re: [LAU] MIDI over ethernet

by Kjetil S. Matheussen

Brett McCoy: > Is there a MIDI over ethernet implementation for Linux? I searched and > couldn't find one except some suggestions to use stuff under WINE. > Midishare. Some work to set up, but it works very well, and it is multi-platform. http://midishare.sf.net

16 years

fun with vowels

by Robert Jonsson

Something the internet found today, just read it on reddit :-) http://www.franciscotufro.com.ar/2009/09/06/algorithmic-composition-using-a… /Robert

16 years

Audio cutoff when playing using dmix

by ruchi sirauthiya

Hi, I am facing audio clipping issues when dmix is used for playback. For the test we used a full scale sine wave tone generated using software and encoded as "Signed 16 bit Little Endian, Rate 44100 Hz, Mono". The audio is getting cut out at a 2-3 second interval during playback. This happens only for a fraction of second but it is consistent. This does not happen when audio is played back on the device directly but only when going through dmix. We have observed that there were differences in audio cutting off when the sampling rate was set to 8000Hz and 32000 Hz in asound.conf. The contents of asound.conf is pasted below for your reference: pcm.dmixer { type dmix ipc_key 2048 ipc_key_add_uid true slave { pcm hw:0 rate 8000 channels 2 } } pcm.asymed { type asym playback.pcm "dmixer" capture.pcm "hw:0,0" } pcm.rateConvert { type plug slave.pcm "asymed" } The configuration given above works fine with dmix in 2.6.29 kernel, I am using kernel 2.6.22.18. I have compared the configuration differences (buffers and periods) with and without using dmix. I tried applying some parameters (like buffer_size, period_time and period_size) which was in use when audio was played directly to the hardware device by aplay using the following command: aplay -D rateConvert --period-time=92879 --buffer-size=65536 --period-size=4096 /mnt/test.wav -vvv -d 5 There was come improvement in the audio cutting off issue but it was not eliminated completely. Given below are the logs for aplay (-vvv) with and without using dmix (with audio cutoff issue on kernel 2.6.22): Without using dmix: [root@OMAP3 tests]# aplay /mnt/test.wav -vvv -d 5 5 Playing WAVE '/mnt/test.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Mono Plug PCM: Hardware PCM card 0 'TWL4030' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 1 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 65536 period_size : 4096 period_time : 92879 tick_time : 7812 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 4096 xfer_align : 4096 start_threshold : 65536 stop_threshold : 65536 silence_threshold: 0 silence_size : 0 boundary : 1073741824 Max peak (4096 samples): 0x00006665 ################ 79% Max peak (4096 samples): 0x00006665 ################ 79% ...Same text repeats 50 times... Max peak (4096 samples): 0x00006665 ################ 79% Max peak (4096 samples): 0x00006665 ################ 79% Using dmix: [root@OMAP3 tests]# aplay -D rateConvert /mnt/test.wav -vvv -d 5 Playing WAVE '/mnt/test.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Mono Plug PCM: Rate conversion PCM (8000, sformat=S16_LE) Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 1 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 22050 period_size : 5512 period_time : 125000 tick_time : 0 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 5512 xfer_align : 5512 start_threshold : 22048 stop_threshold : 22050 silence_threshold: 0 silence_size : 0 boundary : 1445068800 Slave: Route conversion PCM (sformat=S16_LE) Transformation table: 0 <- 0 1 <- 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 1 rate : 8000 exact rate : 8000 (8000/1) msbits : 16 buffer_size : 4000 period_size : 1000 period_time : 125000 tick_time : 0 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 1000 xfer_align : 1000 start_threshold : 4000 stop_threshold : 4000 silence_threshold: 0 silence_size : 0 boundary : 2097152000 Slave: Direct Stream Mixing PCM Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 8000 exact rate : 8000 (8000/1) msbits : 16 buffer_size : 4000 period_size : 1000 period_time : 125000 tick_time : 0 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 1000 xfer_align : 1000 start_threshold : 4000 stop_threshold : 4000 silence_threshold: 0 silence_size : 0 boundary : 2097152000 Hardware PCM card 0 'TWL4030' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 8000 exact rate : 8000 (8000/1) msbits : 16 buffer_size : 16000 period_size : 1000 period_time : 125000 tick_time : 7812 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 1000 xfer_align : 1000 start_threshold : 1 stop_threshold : 2097152000 silence_threshold: 0 silence_size : 2097152000 boundary : 2097152000 Max peak (5512 samples): 0x00006665 ################ 79% Max peak (5512 samples): 0x00006665 ################ 79% ...Same text repeats 36 times... Max peak (5512 samples): 0x00006665 ################ 79% Max peak (5512 samples): 0x00006665 ################ 79% Max peak (5512 samples): 0x00005f1a ############### 74% Please help me fix this issue. Does this occur due to non-optimal settings of the parameters (buffers, periods) in the configuration or is it some other issue with the ALSA library or the kernel? Please ask if any more setup/configuration settings are required for reference. Thanks and Regards, Ruchi Sirauthiya

16 years

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Linux-audio-user September 2009