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7502 discussions

Re: [LAD] mixing while using libao and libsndfile

by Andrea Del Signore

> I've been knocking my head against a wall for more than a year trying to > figure out how to correctly mix two streams of audio while using > libsndfile for input and libao for output. My main requirement is that > I cannot assume anything about the output drivers -- that is, I cannot > depend on the output driver (ALSA, OSS, Sun, etc) being able to do the > mixing for me. Many of my target platforms lack any sort of mixing > services. I need to do this myself. I tried starting a mixer/player > thread that would work in a producer/consumer relationship with one or > two audio file decoder threads. I can play one sound at a time just > fine. When I try to do both, I get distortion followed by a segfault. Hi, not sure if I understood correctly: do you just want to mix N files? Like you I'm learning libsndfile and libao so this is my attempt to mix some audio files: http://pastebin.com/dm7z8b3Z HTH, Andrea P.S. Can someone explain line 88 (I already read the sndfile FAQ)? On Mon, May 16, 2016 at 11:30 PM, < linux-audio-dev-owner(a)lists.linuxaudio.org> wrote: > You are not allowed to post to this mailing list, and your message has > been automatically rejected. If you think that your messages are > being rejected in error, contact the mailing list owner at > linux-audio-dev-owner(a)lists.linuxaudio.org. > > > > ---------- Forwarded message ---------- > From: Andrea Del Signore <sejerpz(a)gmail.com> > To: linux-audio-dev(a)lists.linuxaudio.org > Cc: > Date: Mon, 16 May 2016 21:26:40 +0000 (UTC) > Subject: Re: [LAD] mixing while using libao and libsndfile > On Sun, 15 May 2016 16:34:34 +0000, David Griffith wrote: > > > I've been knocking my head against a wall for more than a year trying to > > figure out how to correctly mix two streams of audio while using > > libsndfile for input and libao for output. My main requirement is that > > I cannot assume anything about the output drivers -- that is, I cannot > > depend on the output driver (ALSA, OSS, Sun, etc) being able to do the > > mixing for me. Many of my target platforms lack any sort of mixing > > services. I need to do this myself. I tried starting a mixer/player > > thread that would work in a producer/consumer relationship with one or > > two audio file decoder threads. I can play one sound at a time just > > fine. When I try to do both, I get distortion followed by a segfault. > > > > So, I'm back to a demo program. What must I do to this program to cause > > it to start playing one audio file, then play another N seconds later? > > > > David Griffith dave(a)661.org > > > > Hi, > > not sure if I understood correctly: do you just want to mix N files? > > I'm a noob with both libsndfile and libao :) > > Here my code: http://pastebin.com/dm7z8b3Z > > HTH, > Andrea > > P.S. > Can someone explain line 88 (I already read the sndfile FAQ)? > > >

9 years, 1 month

mixing while using libao and libsndfile

by David Griffith

I've been knocking my head against a wall for more than a year trying to figure out how to correctly mix two streams of audio while using libsndfile for input and libao for output. My main requirement is that I cannot assume anything about the output drivers -- that is, I cannot depend on the output driver (ALSA, OSS, Sun, etc) being able to do the mixing for me. Many of my target platforms lack any sort of mixing services. I need to do this myself. I tried starting a mixer/player thread that would work in a producer/consumer relationship with one or two audio file decoder threads. I can play one sound at a time just fine. When I try to do both, I get distortion followed by a segfault. So, I'm back to a demo program. What must I do to this program to cause it to start playing one audio file, then play another N seconds later? David Griffith dave(a)661.org ===begin code=== /* * gcc -o mixer mixer.c -lao -lsndfile * */ #include <stdio.h> #include <stdlib.h> #include <stdint.h> #include <string.h> #include <ao/ao.h> #include <sndfile.h> #define BUFFSIZE 512 int playfile(FILE *); int main(int argc, char *argv[]) { FILE *fp1, *fp2; if (argc < 2) { printf("usage: %s <filename>.ogg <filename>.aiff\n", argv[0]); exit(1); } fp1 = fopen(argv[1], "rb"); if (fp1 == NULL) { printf("Cannot open %s.\n", argv[1]); exit(2); } fp2 = fopen(argv[1], "rb"); if (fp2 == NULL) { printf("Cannot open %s.\n", argv[1]); exit(3); } ao_initialize(); playfile(fp1); playfile(fp2); ao_shutdown(); return 0; } int playfile(FILE *fp) { int default_driver; ao_device *device; ao_sample_format format; SNDFILE *sndfile; SF_INFO sf_info; short *shortbuffer; int64_t toread; int64_t frames_read; int64_t count; sndfile = sf_open_fd(fileno(fp), SFM_READ, &sf_info, 1); memset(&format, 0, sizeof(ao_sample_format)); shortbuffer = malloc(BUFFSIZE * sf_info.channels * sizeof(short)); frames_read = 0; toread = sf_info.frames * sf_info.channels; count = 0; default_driver = ao_default_driver_id(); memset(&format, 0, sizeof(ao_sample_format)); format.byte_format = AO_FMT_NATIVE; format.bits = 16; format.channels = sf_info.channels; format.rate = sf_info.samplerate; device = ao_open_live(default_driver, &format, NULL); if (device == NULL) { printf("Error opening sound device.\n"); exit(4); } while (count < toread) { frames_read = sf_read_short(sndfile, shortbuffer, BUFFSIZE); count += frames_read; ao_play(device, (char *)shortbuffer, frames_read * sizeof(short)); } ao_close(device); sf_close(sndfile); } ===end code=== -- David Griffith dave(a)661.org

9 years, 1 month

Jack ring buffer (again)

by Will Godfrey

I'm not sure if this has been covered before... While I understand that generally you can't be certain of writing or reading all bytes in a block of data in one call, what about the specific case where you *always* read and write the same number of bytes and the buffer is an exact multiple of this size. e.g data block is 5 bytes & buffer size is 75 bytes. No I'm not intending to use such an example, I just want to cover worst case :) If that doesn't work, what about the case when you are always working in powers of 2? e.g data block is 16 bytes & buffer size is 1024 bytes. -- Will J Godfrey http://www.musically.me.uk Say you have a poem and I have a tune. Exchange them and we can both have a poem, a tune, and a song.

9 years, 1 month

[LAU] [LAA] Guitarix 0.35.0 released

by Hermann Meyer

The Guitarix developers proudly present Guitarix release 0.35.0 Guitarix is a tube amplifier simulation for jack (Linux), with an additional mono and a stereo effect rack. Guitarix includes a large list of plugins[*] and support LADSPA / LV2 plugs as well. The guitarix engine is designed for LIVE usage, and feature ultra fast, glitch and click free, preset switching, full Midi and/or remote controllable (Web UI not included in the distributed tar ball). This release introduce the new GUI design by Markus Schmidt aka. boomshop Beside that, it comes with a couple of fixes and some new plugins. Also included be the MOD UI's for the LV2 plugins used by the MOD[*] For all changes, please check out the changelog. Please refer to our project page for more information: http://guitarix.sourceforge.net/ Download Site: http://sourceforge.net/projects/guitarix/ Forum: http://guitarix.sourceforge.net/forum/ Please consider visiting our forum or leaving a message on guitarix-developer(a)lists.sourceforge.net <mailto:guitarix-developer@lists.sourceforge.net> regards hermann [*] http://moddevices.com/

9 years, 2 months

Re: [LAD] [FFADO-devel] [PATCH ffado] tests: fix build error due to wrong errno initialization

by Jonathan Woithe

On Fri, Apr 29, 2016 at 09:08:40AM +0200, Jano Svitok wrote: > I tried to compile ffado trunk now (ubuntu 14.04 trusty) and the > compilation failed with undefined errno. > The attached patch fixed that. Thanks. Patch applied as r2607. > I didn't look why the others work and these two files not. The others either don't reference errno directly or include other headers which apparently pull it in. Regards jonathan

9 years, 2 months

After LAC is before LAC

by David Runge

Folks! it has been our pleasure having some of you at c-base the past weekend! It trully was an awesome event. Thank you! Some notes about the event: * all videos here: https://media.ccc.de/c/minilac16 * github association here: https://github.com/linuxaudio * update your information and files for content freeze prior to wiki move before 20160430! http://minilac.linuxaudio.org/ Please watch https://media.ccc.de/v/minilac16-lacisdeadlongliveminilac to get an update on a possible location for next year. The video also serves as explanatory for our proposal of a collective redesign, managed through github. For this to happen, please let me know about your github name, so I can add you! All the best from c-base, David -- David Runge Schreinerstraße 11 10247 Berlin http://sleepmap.de

9 years, 2 months

[ANN] Qtractor 0.7.7 - The Haziest Photon is out!

by Rui Nuno Capela

Hi everybody, On the wrap of the late miniLAC2016(a)c-base.org Berlin (April 8-10) [7][8], where this Yet Same Old Qstuff* (continued) workshop [9] babbling of yours truly (slides, videos[10]) was taken place. There's really one (big) thing to keep in mind, as always: Qtractor [1] is not, never was, meant to be a 'do-it-all' monolith DAW. Quite frankly it isn't a 'pure' modular model either. Maybe we can agree on calling it a 'hybrid' perhaps? And still, all this time, it has been just truthful to its original mission statement--modulo some Qt [2] major version numbers--nb. it started on Qt3 (2005-2007), then Qt4 (2008-2014), it is now Qt5 full throttle. It must have been like start saying uh. this is probably the best dot or rather beta release of them all! Now, Qtractor 0.7.7 (haziest photon) is out! Everybody is here compelled to update. Leave no excuses behind. As for the 'mission statement' coined above, you know it's the same as ever was (and it now goes to eleven years in the making [11]): Qtractor [1] is an audio/MIDI multi-track sequencer application written in C++ with the Qt framework [2]. Target platform is Linux, where the Jack Audio Connection Kit (JACK [3]) for audio and the Advanced Linux Sound Architecture (ALSA [4]) for MIDI are the main infrastructures to evolve as a fairly-featured Linux desktop audio workstation GUI, specially dedicated to the personal home-studio. Website: http://qtractor.sourceforge.net Project page: http://sourceforge.net/projects/qtractor Downloads: http://sourceforge.net/projects/qtractor/files - source tarball: http://download.sf.net/qtractor/qtractor-0.7.7.tar.gz - source package: http://download.sf.net/qtractor/qtractor-0.7.7-25.rncbc.suse.src.rpm - binary packages: http://download.sf.net/qtractor/qtractor-0.7.7-25.rncbc.suse.i586.rpm http://download.sf.net/qtractor/qtractor-0.7.7-25.rncbc.suse.x86_84.rpm Git repos: http://git.code.sf.net/p/qtractor/code https://github.com/rncbc/qtractor.git https://gitlab.com/rncbc/qtractor.git https://bitbucket.org/rncbc/qtractor.git Change-log: - LV2 UI Touch feature/interface support added. - MIDI aware plug-ins are now void from multiple or parallel instantiation. - MIDI tracks and buses plug-in chains now honor the number of effective audio channels from the assigned audio output bus; dedicated audio output ports will keep default to the stereo two channels. - Plug-in rescan option has been added to plug-ins selection dialog (yet another suggestion by Frank Neumann, thanks). - Dropped the --enable-qt5 from configure as found redundant given that's the build default anyway (suggestion by Guido Scholz, thanks). - Immediate visual sync has been added to main and MIDI clip editor thumb-views (a request by Frank Neumann, thanks). - Fixed an old MIDI clip editor contents disappearing bug, which manifested when drawing free-hand (ie. Edit/Select Mode/Edit Draw is on) over and behind its start/beginning position (while in the lower view pane). License: Qtractor [1] is free, open-source Linux Audio [5] software, distributed under the terms of the GNU General Public License (GPL [6]) version 2 or later. References: [1] Qtractor - An audio/MIDI multi-track sequencer http://qtractor.sourceforge.net [2] Qt framework, C++ class library and tools for cross-platform application and UI development http://qt.io/ [3] JACK Audio Connection Kit http://jackaudio.org [4] ALSA, Advanced Linux Sound Architecture http://www.alsa-project.org/ [5] Linux Audio consortium of libre software for audio-related work http://linuxaudio.org [6] GPL - GNU General Public License http://www.gnu.org/copyleft/gpl.html [7] miniLAC, a more compact, community-driven version of the yearly Linux Audio Conference http://minilac.linuxaudio.org [8] c-base.org, Berlin http://c-base.org [9] Yet Same Old Qstuff* (continued) workshop http://minilac.linuxaudio.org/index.php/Workshop#Yet_Same_Old_Qstuff.2A_.28… [10] slides: http://minilac.linuxaudio.org/index.php/File:Lac2016_qstuff_slides.pdf videos: http://media.ccc.de/v/minilac16-yetsameoldqstuff [11] Siskel & Ebert - "This Is Spinal Tap" http://www.youtube.com/watch?v=4xgx4k83zzc See also: http://www.rncbc.org/drupal/node/1033 Enjoy && Have fun. -- rncbc aka Rui Nuno Capela

9 years, 2 months

Adding apps to wiki.linuxaudio.org

by Joel Roth

Hi all, After creating an account for the wiki, it appears that apps are in a separate database from the wiki page itself. I'd like to add an entry for Nama, to the list of DAW software. Can someone put me in touch with the database maintainer? Joel -- Joel Roth

9 years, 2 months

ableton link and live tempo changing

by Perry Nguyen

Hi, I'm Perry, also I'm tatch on linuxmusicians if you've ever been there. I posted this message on LM a while ago but was encouraged by falktx to post this to LAD. so that is what I am doing now (here <https://linuxmusicians.com/viewtopic.php?t=14913> is the original topic). not sure if you’ve read about ableton live 9.5 but one of their planned updates is this thing called ableton link <http://createdigitalmusic.com/2015/11/link-could-change-how-you-play-music-…> that seems to be a convincing inter-device tempo sync. paul davis was present during the unveiling of ableton sync and has an interesting perspective: …The JACK community developed technology back in 2005 or earlier that could > do what Link does (and a whole lot more besides). But the developers failed > to package it, failed to make it usable for anyone other than a small group > of tech-oriented tinkerers, failed to give it an inviting face. In > addition, the developers failed to understand how significant platform is > when trying to create products for people. Designing the most awesome thing > in the world is no help if it doesn’t run on the platforms that people want > to use (for whatever reason). Ableton not only chose OS platforms with huge > numbers of users, but has proceeded to develop its own product-centric > platform that provides compelling reasons for users to want to remain > within its walls. The open source community shouldn’t be trying to follow > or copy what a company like Ableton does, but there is a strong lesson > here: our technology is not as important as we like to think. In any case, one thing that we’ve talked about to various degrees in the past is tempo ramping in JACK, and with the announcement of ableton link I’d like to revisit the topic. http://linuxmusicians.com/viewtopic.php?f=44&t=11681&p=46912 http://linuxmusicians.com/viewtopic.php?f=19&t=2290&p=51449 http://linuxmusicians.com/viewtopic.php?f=1&t=9835&p=30447 JACK does indeed largely do what Link does (and more) but I’ve been having difficulty successfully tempo ramping. Incidentally, danboid recently posted <https://linuxmusicians.com/viewtopic.php?f=1&t=14912> about how he can successfully ramp in MuSE; but I don’t use MuSE nor do I plan to in the near future, and unfortunately my attempts to change tempo dynamically within the greater JACK/modular LAU paradigm have been rather unsuccessful. To clarify my own use-case, I’d like to map an encoder/knob to a tempo control and be able to speed up/slow down the tempo live, e.g. during a performance. *changing tempo as timebase master* For the most part it seems that changing tempo dynamically is possible but it is not well-defined. - seq24 forces itself to be transport master (I know there’s supposed to be a patch but it has still never worked for me ever) and disallows tempo changes while playing - hydrogen sort of allows tempo changes but it will sometimes politely refuse to change (which is bad) - klick can be configured to send out predefined tempo ramps but I haven’t been able to set it as timebase master and actually interact with it (I was trying klick -Ti and klick -To 8080 but it seems -i and -o somehow disregard the -T option) *changing tempo as a slave* - hydrogen is happy to oblige tempo changes but is equally happy to cut off the beginning/end of a bar - qmidiarp follows tempo changes but doesn’t stay in time and forgets to send note-offs I’d appreciate if someone more familiar with the transport spec could chime in on this, I’d like to understand what is missing from a technical perspective (I’m actually not entirely clear on the timebase master/slave relationship either). Going a step further, it seems that ableton link may become the go-to replacement for midi clock in software-based production; it could be beneficial to discuss possible timebase master clock clients that can also communicate using the ableton link protocol. Of course this is contingent on the SDK being compatible with linux, which may be unlikely but they *are* encouraging interested developers to get in touch <https://www.ableton.com/en/link/>. By the same token it would be great to have this for midi clock as well — I know of jack_midi_clock <https://github.com/x42/jack_midi_clock> already but I don’t believe the reverse case exists, though falktx and I briefly discussed something related <https://linuxmusicians.com/viewtopic.php?f=48&t=12755> a while ago. To reiterate, 1. lau apps have to implement tempo ramping in JACK properly 2. a JACK timebase master clock client must be made that can - change tempo dynamically - communicate with ableton link to send/receive tempo information (pending a working SDK for linux (this could also be optional in the app for those who don’t want anything proprietary)) - set tempo given a midi clock input Thoughts? I think tempo syncing is one way we can get JACK to interact more proactively with other software/equipment, and that seems like it could be a great way to reconnect with the rest of the audio community.

9 years, 2 months

Favorite Python3 equivalent of PyJack?

by Jonathan Brickman

I am finding myself motivated to convert from Python 2.7 to Python 3, and I just realized that PyJack, the Python JACK library I have been using, has not been set up for Python 3. Anyone have a favorite JACK library for Python3? -- *Jonathan E. Brickman jeb(a)ponderworthy.com <http://login.jsp/?at=02e47df3-a9af-4cd9-b951-1a06d255b48f&mailto=jeb@ponder…> (785)233-9977* *Hear us at http://ponderworthy.com <http://ponderworthy.com> -- CDs and MP3s now available! <http://ponderworthy.com/ad-astra/ad-astra.html>* *Music of compassion; fire, and life!!!*

9 years, 2 months

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Linux-audio-dev