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From: Peter L. <pl...@in...> - 2005-05-20 14:11:54
|
On Fri, 1 Apr 2005 10:53:22 +0300 (EEST), Kai Vehmanen <kve...@ec...> wrote: > On Wed, 30 Mar 2005, Peter Lutek wrote: > >> i am wondering if it is possible to reset the "over" counters in >> ecasignalview, without stopping the meters. i don't see anything in the >> manpage about a way to do that. > > Unfortunately that's not possible in current versions. Ecasignalview > doesn't allow any user interaction during operation, it just shows the > levels > and that's it. > > Extending ecasignalview with an interactive UI would be a bigger task, > but as a quick hack I added support for resetting the stats by sending a > HUP to the ecasignalview process (i.e. from another console: "killall -v > -HUP ecasignalview"). > > This is available in CVS, and using the snapshot at: > http://ecasound.seul.org/download/snapshots/ecasound-2.4.1-cvs-20050401.tar.gz ok, it's taken me a long time to get around to upgrading to 2.4.1, but i'm there now. resetting the meters works fine, but there's one strange issue i'm noticing (perhaps this is a "feature", which just needs more explanation). if i do the following: STEP1: ecasound -c -r -f:f32,1,44100 \ -a:1 -i:jack_auto,alsa_pcm:capture_1 -o:ch1.wav \ -a:2 -i:jack_auto,alsa_pcm:capture_2 -o:ch2.wav \ -a:3 -i:jack_auto,alsa_pcm:capture_3 -o:ch3.wav \ -a:4 -i:jack_auto,alsa_pcm:capture_4 -o:ch4.wav \ -a:5 -i:jack_auto,alsa_pcm:capture_5 -o:ch5.wav \ -a:6 -i:jack_auto,alsa_pcm:capture_6 -o:ch6.wav \ -a:7 -i:jack_auto,alsa_pcm:capture_7 -o:ch7.wav \ -a:8 -i:jack_auto,alsa_pcm:capture_8 -o:ch8.wav \ -a:9 -i:jack_auto,alsa_pcm:capture_9 -o:ch9.wav \ -a:10 -i:jack_auto,alsa_pcm:capture_10 -o:ch10.wav \ -a:11 -i:jack_auto,alsa_pcm:capture_11 -o:ch11.wav \ -a:12 -i:jack_auto,alsa_pcm:capture_12 -o:ch12.wav \ -a:13 -i:jack_auto,alsa_pcm:capture_13 -o:ch13.wav \ -a:14 -i:jack_auto,alsa_pcm:capture_14 -o:ch14.wav \ -a:15 -i:jack_auto,alsa_pcm:capture_15 -o:ch15.wav \ -a:16 -i:jack_auto,alsa_pcm:capture_16 -o:ch16.wav \ -a:17 -i:jack_auto,alsa_pcm:capture_17 -o:ch17.wav \ -a:18 -i:jack_auto,alsa_pcm:capture_18 -o:ch18.wav STEP2(in another console): ecasignalview -f:16,18,44100 jack_alsa STEP3(in the ecasound-iam console): start STEP4 (from a 3rd console): killall -v -HUP ecasignalview STEP5: (in the ecasound-iam console): stop q ecasound starts writing ch1.wav, ch2.wav, etc. immediately after STEP2, writing blank audio until STEP3, when audio input is supplied, and then writes normally (undisturbed by the HUP) until STEP5. so, i end up with a period of silence at the beginning of all output files, corresponding to the delay between STEP2 and STEP3. shouldn't the ecasound-iam "start" be the trigger for file-writing, rather than the start of a separate ecasignalview process? also, the ecasound-iam "stop" (in my STEP5) stops meter activity in ecasignalview -- the process is still running, and still connected in jack, but the meters stop responding. so, overall, there's some interaction between the ecasound-iam process and the ecasignalview process. perhaps this is intended, but it seems strange to me, and makes usage of ecasound with metering ackward. should this interaction be regarded as bugs? if not, perhaps you could explain your rationale for the behaviour. thanks! -p (sorry for the double message, kai.... i mistakenly sent one from an address not subscirbed to ecasound-list) |
From: Julien C. <ju...@c-...> - 2005-05-16 08:28:49
|
Hi Martin! Depends if you're on the right way. If you just want to play a piece to specific channels constantly. If so, this is possible. You will need the right .asoundrc (alsa configuration), you can get help from the linux-audio-user list or the alsa-user list there. If you created a device in the .asoundrc, you can access it like this: ecasound -i file.wav -o alsa,my_device [other options as you choose] If you want to have it moving, I would suggest using tool more suited to it: csound (nasty language at first, but effective and powerful), clm (commonLispMusic). In the case of more complicated work, keep a look out for them and other synthesizers, audio/video modeling apps. I hope that helps! Kindest regards 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 |
From: Martin S. <sch...@gm...> - 2005-05-13 08:40:43
|
Hi List, I am a Linux audio newbie and I have the following problem: For a VR installation I need a software that can be triggered in realtime to play WAVs on a specified channel on a soundcard. At the moment I work with a standard stereo card, but we will purchase a hammerfall card shortly. Am I on the right track with ecasound? How can I access specific channels? Do I need Jack? If yes, how do I assign a single channel as target? Sorry if this was answered elsewhere, I didn't find anything in the documentation. Thank you! Martin S. |
From: Eric D. R. <er...@zh...> - 2005-04-29 11:59:35
|
Kai Vehmanen wrote: > On Wed, 27 Apr 2005, Eric Dantan Rzewnicki wrote: >> There is now copious output from ecasound when using .ewf files. The >> follwing lines are repeated continuously with Global position and >> child-pos updated with each iteration: > Oops, sorry, that code wasn't supposed to go into the snapshot. I've now > taken the snapshot version offline. I've been using this debug stuff to > trace the ewf-problems Julien reported a while ago. This work is still > ongoing... That's what I figured. No worries. :) -Eric Rz. |
From: Kai V. <kve...@ec...> - 2005-04-28 20:24:38
|
On Wed, 27 Apr 2005, Eric Dantan Rzewnicki wrote: > There is now copious output from ecasound when using .ewf files. The > follwing lines are repeated continuously with Global position and > child-pos updated with each iteration: Oops, sorry, that code wasn't supposed to go into the snapshot. I've now taken the snapshot version offline. I've been using this debug stuff to trace the ewf-problems Julien reported a while ago. This work is still ongoing... -- http://www.eca.cx Audio software for Linux! |
From: Kai V. <kve...@ec...> - 2005-04-28 20:20:24
|
On Thu, 28 Apr 2005, Tommi Sakari Uimonen wrote: > (eca-controller) Loaded chainsetup from file > "/tmp/ecasound-tuimonen/cs-edit-tmp-5455-1.ecs". > Warning: DBC_REQUIRE failed - "is_enabled() == true", eca-chainsetup.cpp, > 1951. > > After the las message, ecasound exits. This is probably a NPTL-issue - clip from BUGS: --cut-- - certain combinations of glibc and linux-2.6 kernels cause the 'cs-edit' EIAM command to fail (ecasound exits back to shell after the command); as a workaround, set the LD_ASSUME_KERNEL=2.2.5 environment variable before running ecasound; see http://people.redhat.com/drepper/assumekernel.html for background information on LD_ASSUME_KERNEL --cut-- -- http://www.eca.cx Audio software for Linux! |
From: Eric D. R. <er...@zh...> - 2005-04-27 23:48:38
|
Commenting out line 126 of libecasound/audioio-ewf.cpp stops the copious output: //dump_child_debug(); and now my scripts are happy again. Eric Dantan Rzewnicki wrote: > Hi Kai, > > There is now copious output from ecasound when using .ewf files. The > follwing lines are repeated continuously with Global position and > child-pos updated with each iteration: > Global position (in samples): 147456 > child-pos: 147456 > child-offset: 0 > child-startpos: 0 > child-length: 1132297 > child-active: 1 > > This output causes a problem for my pyeca eci scripts. The eci parser > buffer overflows. I've put a simple test case, .ewf and .wav file here: > http://zhevny.com/debug/ > running debug2.py causes the following error messages: > > (ecasoundc_sa) Error='read() error', cmd='engine-status' last_error='' > cmd_cnt=8 last_cnt=7. > > *********************************************************************** > * Message from libecasoundc: > * > * Connection to the processing engine was lost. Check that ecasound > * is correctly installed. Also make sure that ecasound is either > * in some directory listed in PATH, or the environment variable > * 'ECASOUND' contains the path to a working ecasound executable. > *********************************************************************** > > > (ecasoundc_sa) Error='sync error', cmd='stop' last_error='' cmd_cnt=9 > last_cnt=7. |
From: Eric D. R. <er...@zh...> - 2005-04-27 23:29:25
|
Hi Kai, There is now copious output from ecasound when using .ewf files. The follwing lines are repeated continuously with Global position and child-pos updated with each iteration: Global position (in samples): 147456 child-pos: 147456 child-offset: 0 child-startpos: 0 child-length: 1132297 child-active: 1 This output causes a problem for my pyeca eci scripts. The eci parser buffer overflows. I've put a simple test case, .ewf and .wav file here: http://zhevny.com/debug/ running debug2.py causes the following error messages: (ecasoundc_sa) Error='read() error', cmd='engine-status' last_error='' cmd_cnt=8 last_cnt=7. *********************************************************************** * Message from libecasoundc: * * Connection to the processing engine was lost. Check that ecasound * is correctly installed. Also make sure that ecasound is either * in some directory listed in PATH, or the environment variable * 'ECASOUND' contains the path to a working ecasound executable. *********************************************************************** (ecasoundc_sa) Error='sync error', cmd='stop' last_error='' cmd_cnt=9 last_cnt=7. This does not happen if I use the .wav file dirrectly. Let me know if there is anything further I can do to help debug this, or if I'm just doing (or not doing) something stupid. Thanks, Eric Rz. Current settup: debian testing (sarge) 2.6.11.7-audiobox-0 includes alsa-drivers 1.0.8 realtime-0.8.5 lsm (insmod'ed in /etc/init.d/bootmisc.sh) insmod /usr/src/realtime/realtime.ko gid=29 alsa-1.0.8 (lib, envy24control, utils) libsndfile-1.0.12pre6 from tar.gz libsamplerate-0.1.2 from tar.gz jack-0.99.61 cvs 2005-0427-0755 ./configure --enable-capabilities --enable-optimize --with-default-tmpdir=/dev/shm --disable-portaudio --disable-coreaudio --disable-oss ecasound-2.4.2-cvs-20050425 ./configure --enable-pyecasound=c --disable-oss --disable-arts --with-largefile ardour-0.9beta29 asus a7v8x-x ATHLON xp 2800+ (2071.203 MHz) 1GB PC2700 RAM 2GB swap /dev/hda1 12GB /dev/hda2 / (actually a 40G disk) ext2 160GB /dev/hdc1 /mnt/audio/ ext2 200GB /dev/hdd1 /mnt/audio/inprogress/studio_space/ ext2 ati rage 128 pro ultra tf (onboard via8235 -- disabled) ice1712 m-audio delta-66 w/omni i/o ymfpci guillemot maxisound fortissimo -- used for midi only novation remote 25 connected to ymfpci |
From: Tommi S. U. <tui...@cc...> - 2005-04-27 21:37:15
|
- [ Chainsetup created (file) ] -------------------------------------------- - [ Controller/Processing stopped (cond) ] --------------------------------- (eca-chainsetup-parser) Buffering mode 'nonrt' selected. (eca-chainsetup-parser) Setting chainsetup name to "mp3". (eca-chainsetup-parser) Updating outputs (rw-mode). (eca-chainsetup-parser) Ignoring xruns during processing. (eca-chainsetup-parser) Enabling precise-sample-rates with OSS audio devices. (eca-controller) Loaded chainsetup from file "/tmp/ecasound-tuimonen/cs-edit-tmp-5455-1.ecs". Warning: DBC_REQUIRE failed - "is_enabled() == true", eca-chainsetup.cpp, 1951. After the las message, ecasound exits. Tommi |
From: Justin R. <myr...@co...> - 2005-04-26 01:09:58
|
This is something I have contemplated for myself, but as Kai stated, you'll need to find an app for that (I haven't found one, and would rather spend time making music than googling). The perl module for ecasound does some pretty cool stuff. You may want to give that a try and see what you come up with. Writing a script for it is pretty easy. I was going to use it for a solo performance, playing some songs back to back, with a little intermittent pause in between. I don't know if there are any Perl modules that would be able to send MIDI interrupts accordingly to, say, make a program change on an effects rack? But, I don't see why the perl module for ecasound wouldn't be able to make the appropriate effect changes for you. On Sun, 2005-04-24 at 23:15 +0300, Kai Vehmanen wrote: > Hi, > > On Wed, 16 Feb 2005, Sean Edwards wrote: > > > I am looking for a way to synchronize sound file play > > back with effect changes to a live guitar. I am > [...] > > midi complexity. In other words, when the effects > > settings for the guitar change during the play back of > > the sound file, it should be done automatically and at > > the right time. Also, the solution needs to be > [...] > > Ecasound has all the requirements: console based, > > real-time effects, sound file playback. I was > > wondering if anybody has put the pieces together like > > this before, or if it is possible. Do I need some > > other pieces to connect with something like Jack? > > as demonstrated by the lack of responses, I guess nobody has yet put this > kind of system together. But it is certainly doable. I would say writing a > small custom app (maybe with Python, Ruby, Perl or some other > nice interpreted language), and using Ecasound's ECI API [1] to control > audio processing, is the best way to go. > > [1] http://eca.cx/eci > |
From: Kai V. <kve...@ec...> - 2005-04-24 20:26:17
|
Hello, On Mon, 7 Mar 2005, Jack O'Quin wrote: > Interpreting the transport state transition from Rolling to Stopped as > an EOF certainly makes sense in many situations. But, I recall trying > to do something similar to Olivier, repeatedly stopping and starting a > non-interactive ecasound session. That would be nice to have (as an > option, perhaps). yup, I agree. I'll try to come up with some option to control this behaviour in the next release. > This raises the question of how to terminate a non-interactive > ecasound. Olivier's suggestion (Ctrl-C) seems OK to me. ? :o Ctrl-c has always been a supported way of terminating ecasound. It will result in a controlled shutdown (for example wav-file headers are updated before closing, connections to servers are closed, ...). So IOW if it doesn't work, it's a bug => let me know about the details. -- http://www.eca.cx Audio software for Linux! |
From: Kai V. <kve...@ec...> - 2005-04-24 20:20:14
|
On Fri, 25 Feb 2005 jef...@ag... wrote: > It is possible to set-up a gated ecasound record chain where the > threshold is set to zero? I am trying to sample-sync 6 tracks sourced > from my external HD recorder. The tracks are downloaded to the computer [...] > ecasound recieves it's first non-zero sample, I want it to start saving > the data to a file. Using -ge:0,0,peak doesn't seem to work. With this > setting ecasound begins saving data immediately. Hmm, -ge:0,0 should work. There are too possible problems: 1) the input samples before actual audio data are not zeros ... you could try with -ge:0.1,0 instead and see if it works better 2) -ge is block based (= not sample-accurate), so when the gate opens, all the samples inside that block are passed through... you can use the -b:xxx option to set the blocksize => with a really small blocksize like -b:16 the worst-case error is just a couple of samples PS It is not impossible to make -ge sample-accurate, but can't promise you anything about the implementation schedule. -- http://www.eca.cx Audio software for Linux! |
From: Kai V. <kve...@ec...> - 2005-04-24 20:12:50
|
Hi, On Wed, 16 Feb 2005, Sean Edwards wrote: > I am looking for a way to synchronize sound file play > back with effect changes to a live guitar. I am [...] > midi complexity. In other words, when the effects > settings for the guitar change during the play back of > the sound file, it should be done automatically and at > the right time. Also, the solution needs to be [...] > Ecasound has all the requirements: console based, > real-time effects, sound file playback. I was > wondering if anybody has put the pieces together like > this before, or if it is possible. Do I need some > other pieces to connect with something like Jack? as demonstrated by the lack of responses, I guess nobody has yet put this kind of system together. But it is certainly doable. I would say writing a small custom app (maybe with Python, Ruby, Perl or some other nice interpreted language), and using Ecasound's ECI API [1] to control audio processing, is the best way to go. [1] http://eca.cx/eci -- http://www.eca.cx Audio software for Linux! |
From: Kai V. <kve...@ec...> - 2005-04-24 20:10:08
|
On Fri, 22 Apr 2005, Julien Claassen wrote: > I just wanted to take the chance to announce again my toalk on the > linux-audio-conference 2005 on ecasound. My talk will be on Saturday I think > 10AM UTC (11am German time). It's in the media theatre. This was great, many thanks for (again) presenting Ecasound at the event! For those who missed the show, a video of the presentation is available at: http://www.linuxdj.com/audio/lad/contrib/zkm_meeting_2005/video/julien_claasen-linux_textbased_studio.theora.ogg Tips on how to view these clips (Vorbis Theora coding): http://footils.org/cms/pydiddy/wiki/WatchingStreams I tested vlc, helix-player, xine and totem, and all worked fine. Julien's slides ar available at: https://sourceforge.net/project/showfiles.php?group_id=93153&package_id=150523 > N.B.: Fro stream-breakdowns they chose making coffee by Kai... :-) This track hasn't been available for download for a long time, so you can imagine my suprise when I tuned to the lac2005 audio-stream on Friday and heard the familiar sounds of my old coffee machine! :) -- http://www.eca.cx Audio software for Linux! |
From: Dan W. <da...@mi...> - 2005-04-23 15:55:28
|
I wrote a ulaw encoder, and I'd be more than happy to donate it to ecasound. Are you interested? Dan |
From: Kai V. <kve...@ec...> - 2005-04-23 15:13:33
|
Hi, On Fri, 25 Feb 2005, Dan Weber wrote: > How do I do ulaw raw output from ecasound? unfortunately ecasound doesn't support ulaw (nor alaw). I recommend sox [1] for converting to ulaw format. [1] http://sox.sourceforge.net/ -- http://www.eca.cx Audio software for Linux! |
From: Free L. <fre...@gm...> - 2005-04-23 14:23:08
|
Hi Kai, Thank you for your response. On 4/17/05, Kai Vehmanen <kve...@ec...> wrote: > On Fri, 15 Apr 2005, Free Lunch wrote: >=20 > > Something bad happened at one point.. My first WAV ended at 471707692 > > bytes. Fortunately, I ran everything under wrapper scripts to > > automatically re-start any applications that failed (jackd, ecasound, > > ecasignal). Unfortunately, I didn't log the output from the software > > so there isn't really any diag info available. >=20 > I recommend running long test runs with the same exact setup, and try > to see if you can recreate the problem. If jackd is run with strict > settings, the system has to be fairly well tuned so that it runs without > glitches for long periods of time. On the bright side, once tuned, jackd > runs does run reliably. I haven't done any jackd tuning yet. I was running defaults. The system config was quite stripped. I have done some testing but need to put together a better test rig. One challenge to accurately reproducing real world recording involves file size limitations - limited laptop disk space. The other involves WAV file size limitations. Both of those could be solved if ecasound had a feature to automatically start a new recording file after a specific file size. I don't believe it has that feature at this time? Sure, I could send the output to /dev/null or null but that wouldn't generate the disk I/O interrupts, etc. > You should also watch out for system events: heavy X screensavers, > logrotated, updatedb and etc cron runs. All good points. I wasn't running X. > A known design limitation of JACK is that changing connections can cause > glitches to audio (in the same way as plugging analog mixer cables can > cause pops ;)). Good to know. > > I would still like to run ecasound and ecasignalview simultaneously > > without Jack, if that is possible. Simpler is better in these cases. >=20 > You can use the ALSA dsnoop plugin: I have been playing with this a bit but need to play with .asoundrc settings a bit more. I just started working with an Edirol UA-5 so I have a lot of testing to do. I must say that the ecasignalview metering is *very* good. The quick update and stats tracking catches clipping that most monitors would miss. While I know that the update rate is adjustable, it would be nice if there was a way to easily reset the peak value and maybe set a 'peak hold'. It might also be neat to have a 'log' of sorts: 10:18:24 Ch-01 1.0000 peak That way, when you look over and see that you clipped, it would be possible to check a tail'd log to see when, etc. Thanks again, FL |
From: Kai V. <kve...@ec...> - 2005-04-23 13:54:11
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On Tue, 15 Feb 2005, Cassettes2CDs wrote: > I do bunches and bunches of stereo 32-bit recording > with ecasound and my M-Audio Delta 1010LT's. I > currently do up to 8 stereo pairs at once, but don't > need them to be -sychronized- with each other at all. So are you using ALSA/asounrc and the multi-plugin, or the 4Front OSS drivers, to provide the stereo pairs...? > The recording process is very fragile, though, and if > the computer does anything disk-y, I drop frames. Hmm, have you compared recording invidual stereo pairs vs recording all channels with single ecasound instance? The latter one should not be fragile even on a not-so-well-tuned system. > ecasound -z:db -r -b 8192 -f 32,2,44100,i -i /dev/dsp5 > -el valve,0.015,0.015 -f 32,2,44100,i -o filename.raw [...] > So my -z is "db," my -b is "8192," and I don't even > use -B. Basicly -B option selects the profile (i.e. which set of default parameters to use for all buffering related options). The other options (-z:db, -b, etc) then tune individual parameters. A good intro to this topic can be found at: http://eca.cx/ecasound-list/2001/10/0020.html A link to above message is mentioned on the tutorials page at: http://eca.cx/ecasound/Documentation/tutorials.html Now as for reliability, -z:db protects against disk i/o scheduling problems, while -b and -z:intbuf/-z:nointbuf help against kernel process scheduling and interrupt latency related problems. For very reliable recording, the following set of option should provide good results in most cases: ecasound -z:db,500000 -r -b:1024 -z:intbuf -f:32,2,44100,i -i /dev/dsp5 -o foo.raw With -z:intbuf, the value of -b:xxx doesn't have a big impact anymore, as then ecasound will in any case try to utilize maximum amount of sound device buffering. With -z:intbuf, -b:xxx then just controls how often interrupts are raised (1024-4096 are reasonable values, many soundcards do not support higher values). I've also increased the disk i/o buffer to 50000 frames (default 100000). With 44100 srate, that's about +5secs of buffering for "foo.raw" and other disk objects in the setup, which is quite a lot. But I also want to emphasize that normally you shouldn't need to specify any of these options. Ecasound should be able to guess the right options to use for most chainsetups. -- http://www.eca.cx Audio software for Linux! |
From: Kai V. <kve...@ec...> - 2005-04-23 13:25:05
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Hello, On Fri, 28 Jan 2005 wi...@ph... wrote: > Even with several inputs, the resultant ecasound mixed file does not clip > if none of the inputs clip. > 1)Without extra parameters, does ecasound average the levels for the > several inputs in mixes? yes, this has been the behavour since the beginning. In version 2.4.0 (and newer), you can alternatively choose sum-mixmode with '-z:mixmode,sum'. Averaging is still the default mixing policy. > 2)Originally we were only going to use ecasound for recording(first app. > that didn't clip :-)) Now we are planning to use ecasound for more > elaborate mixes. It seems after we mix we need to run ecanormalize as the > output file can sometimes have fairly low levels. I could well be doing > something fundamentally wrong, and we do not apply gain/pan individual > channels as many on this list seem to. However, before I spend a lot of > time playing with mixes, any ideas? Yes, as mixing is done by averaging, the resulting amplitude can be lower (especially if you are mixing channels with no overlapping audio). You can either use the sum-mixmode, or add amplify-effects to the chains ("-ea:200" to double amplitude). See the "Mixing" examples at http://eca.cx/ecasound/Documentation/examples.html ... -- http://www.eca.cx Audio software for Linux! |
From: Julien C. <ju...@c-...> - 2005-04-22 14:07:13
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Hi all! I just wanted to take the chance to announce again my toalk on the linux-audio-conference 2005 on ecasound. My talk will be on Saturday I think 10AM UTC (11am German time). It's in the media theatre. You should turn into the streams these days anyway, find them at: http://lac2005.zkm.de (choose automatic streaming directory). Btw.: All music played between the talks and before the talks is done with ecasound. N.B.: Fro stream-breakdowns they chose making coffee by Kai... :-) Kindest regards from Karlsruhe 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 |
From: Stephen S. <lp...@pa...> - 2005-04-21 07:36:24
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Just a note to say that Ecasound 2.4.1 resolved a number of minor issues I've had and it just purrs on my antique 200MHz w/Slackware. Also, it's been some time since I could compile it at all, which wasn't really an issue, since 2.3.4 still worked OK. But specifically, the issue of file names with spaces is now fixed, and for whatever reason, I seem to be able to do a lot more with this release, using less CPU power. Great work, and thank you! :) Steve |
From: Kai V. <kve...@ec...> - 2005-04-17 16:44:32
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On Sun, 17 Apr 2005, Junichi Uekawa wrote: >> I would like to record from alsa with ecasound while simultaneously >> monitoring levels with ecasignalview. > I have a gut feeling that you could do it with alsa, not > something within ecasound. Yups, see my previous examples with asoundrc example. This is quite nice way to achieve simultaneous recording+monitoring. > I think you should be able to do it through jack with something > like: [...] > jackd -d dummy & > ecasound -a rec,mon -i alsahw,1,0,0 -a rec -o file.wav -a mon -o jack_generic:monitor > ecasignalview jack_generic:monitor > > But apparently ecasignalview doesn't like complex names. Hmm, this is a clever idea. :) But, but, you'd still have the problem that JACK could kick the clients (=ecasound) out when disconnecting (ecasignalview). As for the syntax, it should be "jack_generic,monitor". Also, the "jack_generic" variant just publishes the ports, but doesn't connect anything. So a better one would be: ecasound -a rec,mon -i alsahw,1,0,0 -a rec -o file.wav -a mon -o jack -G:jack,ecarec ecasignalview jack_auto:ecarec null The first one will appear as client "ecarec", with ports "out_1", "out_2", etc. Ecasignalview will automatically connect to the "ecarec" client and its ports. -- http://www.eca.cx Audio software for Linux! |
From: Kai V. <kve...@ec...> - 2005-04-17 12:35:51
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On Fri, 15 Apr 2005, Free Lunch wrote: > Something bad happened at one point.. My first WAV ended at 471707692 > bytes. Fortunately, I ran everything under wrapper scripts to > automatically re-start any applications that failed (jackd, ecasound, > ecasignal). Unfortunately, I didn't log the output from the software > so there isn't really any diag info available. I recommend running long test runs with the same exact setup, and try to see if you can recreate the problem. If jackd is run with strict settings, the system has to be fairly well tuned so that it runs without glitches for long periods of time. On the bright side, once tuned, jackd runs does run reliably. You should also watch out for system events: heavy X screensavers, logrotated, updatedb and etc cron runs. These won't necessarily cause problems to audio, but still they cause unnecessary extra load that you don't when doing audio work. I once had updatedb running during a JACK demo at one event, and even though window drawing was _very_ slow, audio played without glitches the whole time! Of course during the demo no idea on what in the world was happening with the system - I had just tested the whole demo on the same hw the day before. :) > One serious glitch occurs when I terminate ecasignalview with an > interrupt. It often causes ecasoundview to also terminate (and my > recording!). I assume that is not expected behavior? I think there > was a message about 'audio chain broken'. I will try and reproduce. A known design limitation of JACK is that changing connections can cause glitches to audio (in the same way as plugging analog mixer cables can cause pops ;)). Other client connections should not be broken, but in practise it often happens - especially other clients connected to same ports as the ones you've disconnected, might need to be reconnected. There have been some proposal how to address these problems, so situation might improve in later JACK releases (=> but not before 1.0). > I would still like to run ecasound and ecasignalview simultaneously > without Jack, if that is possible. Simpler is better in these cases. You can use the ALSA dsnoop plugin: http://www.alsa-project.org/alsa-doc/doc-php/asoundrc.php -> "Software mixing" -> "dsnoop" Copy and paste the "pcm.mixin" definition to your ~/.asoundrc and then you can do: ecasound -i alsa,mixin -o foo.wav ecasignalview alsa,mixin null At least on my system, I don't get any glitches when I start and stop multiple apps using the 'mixin' device. In addition to the above link, another good resource to learn about asoundrc is: http://alsa.opensrc.org/index.php?page=.asoundrc I'm no expert with this stuff, so I recommend subscribing to alsa-user mailing list if you have any tricky questions about asoundrc. :) -- http://www.eca.cx Audio software for Linux! |
From: Junichi U. <da...@ne...> - 2005-04-17 11:26:53
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Hi, > I would like to record from alsa with ecasound while simultaneously > monitoring levels with ecasignalview. > > But they won't share. Is this possible? > > I couldn't find an example of this in the docs or via google. That would be very nice; my current use-scenarios would benefit from that. I have a gut feeling that you could do it with alsa, not something within ecasound. I think you should be able to do it through jack with something like: jackd -d dummy & ecasound -a rec,mon -i alsahw,1,0,0 -a rec -o file.wav -a mon -o jack_generic:monitor ecasignalview jack_generic:monitor But apparently ecasignalview doesn't like complex names. regards, junichi |
From: Kai V. <kve...@ec...> - 2005-04-17 09:24:13
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On Fri, 15 Apr 2005, Lars Bj=F8rndal wrote: > Is it possible to do this from the command line, without changing > anything in the ecasoundrc file? I've tried the -f: option, but I > still get 2 channels. Unfortunately that's not supported. You have to manually set up pipes to do that: ecasound -f:16,2,44100 -i foo.mp3 -f:16,1,44100 -o stdout -erm:1 |lame -b 1= 28 -s 44.1 -x -m m - foo-mono.mp3 You can also try directly with lame (not tested): lame -b 128 -s 44.1 -x -m m --mp3input -a foo.mp3 foo-mono.mp3 But note that you will lose in quality by doing this, as ecasound (I'm not= =20 100% sure about lame, but I think it will re-encode) has to decode and=20 then re-encode the audio data =3D> information is lost. I recommend using= =20 non-lossy codecs if you need to process your audio files. Converting to a= =20 lossy format (such as mp3), should always be the very last step. --=20 http://www.eca.cx Audio software for Linux! |