Linux-audio-dev

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7516 discussions

midi2tcp and tcp2midi, working prototype

by Jonathan E. Brickman

It is working as here: https://github.com/ponderworthy/midi2tcp and quite well. Haven't come up with a single reason to make it more complicated, beyond much more careful timeout handling. It will need to be able to destroy its own socket and recreate et cetera, to be durable with marginal wifi. And performance is not very good, midi2tcp takes 3-5% of one core and tcp2midi takes 6-8% of another on this i3- 3110 at 2.4GHz. Am not sure what best to do about the performance. Will probably try converting Python to binary, but have read over and over again that that doesn't do so very much. I also had to rename it back to TCP after re-reading the Mido library docs yet again, it is using rtpmidi backend but it says it's TCP, not UDP, and it is using a connected socket which means TCP I do believe. I had thought RTP-MIDI was UDP? I wonder if judicious use of UDP would improve performance by a substantial amount? -- Jonathan E. Brickman jeb(a)ponderworthy.com (785)233-9977 Hear us at ponderworthy.com -- CDs and MP3 available! Music of compassion; fire, and life!!!

7 years, 2 months

MIDI-2-TCP, TCP-2-MIDI

by Jonathan E. Brickman

I need lossless JACK MIDI networking outside of JACK's built-in networking, and not multicast unless someone can tell me straightforwardly how to get multicast (qmidinet) to run within localhost as well as outside it. Thus I am thinking of trying my hand at using the Mido library to bridge JACK MIDI and TCP. I have never done this sort of coding before, programmatorially I am mostly a deep scripting guy, Python-heavy with a bunch of Bash on Linux, Powershell- heavy on Windows of late, with a pile of history on back in Perl on both and VBA on Windows. Anyone have hints...suggestions...alternatives...a best or better starting place? Right now I don't want the applets to do GUI at all, I just want them to sit quietly in xterms, on JACK servers, keeping connection, and passing MIDI data to and fro, as other processes and devices bring it. -- Jonathan E. Brickman jeb(a)ponderworthy.com (785)233-9977 Hear us at ponderworthy.com -- CDs and MP3 available! Music of compassion; fire, and life!!!

7 years, 2 months

Berlin Linux Audio meeting @ c-base 2018-09-11

by Daniel Swärd

Hi all. Next meeting at c-base is on Tuesday 2018-09-11. To avoid cross-posting too much on the mailing lists I'd like the follow-up discussion to happen in the thread to the invitation mail I sent to the linux- audio-user list. There's also a bit more content in that mail than a redirect... ;-) Cheers /Daniel

7 years, 2 months

spectmorph-0.4.1

by Stefan Westerfeld

spectmorph-0.4.1 has been released. Overview of Changes in spectmorph-0.4.1: ---------------------------------------- * macOS is now supported: provide VST plugin for macOS >= 10.9 * Include instruments in source tarball and packages * Install instruments to system-wide location * New Instruments: Claudia Ah / Ih / Oh (female version of human voice) * Improved tools for instrument building - support displaying tuning in sminspector - implement "smooth-tune" command for reducing vibrato from recordings - minor encoder fixes/cleanups - smlive now supports enable/disable noise * VST plugin: fix automation in Cubase (define "effCanBeAutomated") * UI: use Source A / Source B instead of Left Source / Right Source * UI: update db label properly on grid instrument selection change * Avoid exporting symbols that don't belong to the SpectMorph namespace * Fix some LV2 ttl problems * Fix locale related problems when using atof() * Minor fixes and cleanups What is SpectMorph? ------------------- SpectMorph is a free software project which allows to analyze samples of musical instruments, and to combine them (morphing). It can be used to construct hybrid sounds, for instance a sound between a trumpet and a flute; or smooth transitions, for instance a sound that starts as a trumpet and then gradually changes to a flute. SpectMorph ships with many ready-to-use instruments which can be combined using morphing. SpectMorph is implemented in C++ and licensed under the GNU LGPL version 3 Integrating SpectMorph into your Work ------------------------------------- SpectMorph is currently available for Linux and Windows users. Here is a quick overview of how you can make music using SpectMorph. - VST Plugin, especially for proprietary solutions that don't support LV2. (Available on Linux and 64-bit Windows) - LV2 Plugin, for any sequencer that supports it. - JACK Client. - BEAST Module, integrating into BEASTs modular environment. Note that at this point, we may still change the way sound synthesis works, so newer versions of SpectMorph may sound (slightly) different than the current version. Links: ------ Website: http://www.spectmorph.org Download: http://www.spectmorph.org/downloads There are many audio demos on the website, which demonstrate morphing between instruments. -- Stefan Westerfeld, Hamburg/Germany, http://space.twc.de/~stefan

7 years, 2 months

Re: [LAD] ?= Updat

by bill-auger

Spam detection software, running on the system "mail1.linuxaudio.cyso.net", has identified this incoming email as possible spam. The original message has been attached to this so you can view it or label similar future email. If you have any questions, see @@CONTACT_ADDRESS@@ for details. Content preview: yes that was a not a great analogy - my example was only meant to be general advice for any arbitrary dependency - it did not fit well for this case where that dependency is part of the tool-chain - indeed, you are correct that the very purpose of makefiles is to abstract over system-specifics in the tool-chain (and pkg-config itself is for abstracting system-specific library filenames) - if the two pkg-config binaries had different filenames then it would perhaps call for a new makefile variable like: $PKGCONF [...] Content analysis details: (5.5 points, 5.0 required) pts rule name description ---- ---------------------- -------------------------------------------------- 2.5 RCVD_IN_SORBS_HTTP RBL: SORBS: sender is open HTTP proxy server [75.138.187.221 listed in dnsbl.sorbs.net] 2.4 RCVD_IN_SORBS_SOCKS RBL: SORBS: sender is open SOCKS proxy server 1.1 DATE_IN_PAST_03_06 Date: is 3 to 6 hours before Received: date -0.5 AWL AWL: Adjusted score from AWL reputation of From: address

7 years, 2 months

Re: [LAD] ?==?utf-8?q? Update

by bill-auger

Spam detection software, running on the system "mail1.linuxaudio.cyso.net", has identified this incoming email as possible spam. The original message has been attached to this so you can view it or label similar future email. If you have any questions, see @@CONTACT_ADDRESS@@ for details. Content preview: On Tue, 28 Aug 2018 16:48:40 +0200 Hermann Meyer wrote: > Using pkg-config means that it works on any system, even older ones, using pkg-config means that it works on any system that has pkg-config installed - using pkgconf means that it works on any system that has pkgconf installed - and either of pkg-config and pkgconf can be installed on any system regardless of what the distro declares as the "official" implementation [...] Content analysis details: (5.4 points, 5.0 required) pts rule name description ---- ---------------------- -------------------------------------------------- 0.0 BAD_ENC_HEADER Message has bad MIME encoding in the header 2.5 RCVD_IN_SORBS_HTTP RBL: SORBS: sender is open HTTP proxy server [75.138.187.221 listed in dnsbl.sorbs.net] 2.4 RCVD_IN_SORBS_SOCKS RBL: SORBS: sender is open SOCKS proxy server 1.1 DATE_IN_PAST_03_06 Date: is 3 to 6 hours before Received: date 0.1 DKIM_SIGNED Message has a DKIM or DK signature, not necessarily valid -0.1 DKIM_VALID Message has at least one valid DKIM or DK signature -0.1 DKIM_VALID_AU Message has a valid DKIM or DK signature from author's domain -0.5 AWL AWL: Adjusted score from AWL reputation of From: address

7 years, 2 months

Updates

by Fons Adriaensen

Hello all, Updates are available at <http://kokkinizita.linuxaudio.org/linuxaudio/downloads/index.html> for all of the following: aeolus-0.9.7 ambdec-0.7.1 clthreads-2.4.2 clxclient-3.9.2 ebumeter-0.4.2 jaaa-0.9.2 jack_delay-0.4.2 jack_utils-0.0.1 japa-0.9.2 jconvolver-1.0.2 jkmeter-0.6.5 jmatconvol-0.3.3 jmeters-0.4.4 jnoisemeter-0.2.2 octofile-0.3.2 tetraproc-0.8.6 yass-0.1.0 zita-ajbridge-0.8.2 zita-alsa-pcmi-0.3.2 zita-at1-0.6.2 zita-bls1-0.3.3 zita-convolver-4.0.2 zita-dpl1-0.3.3 zita-jclient-0.4.2 zita-lrx-0.1.2 zita-mu1-0.3.3 zita-njbridge-0.4.4 zita-resampler-1.6.2 zita-rev1-0.2.2 For most of them this is just small bug fixes, general maintenance, and above all a systematic cleanup of the Makefiles. Also the jacktools python package (presented at LAC 2018) is available now. I actually did upload the files just before the conference, but forgot to update index.html... Ciao, -- FA

7 years, 2 months

PipeWire

by Wim Taymans

Hi everyone, I'm Wim Taymans and I'm working on a new project called PipeWire you might have heard about [1]. I have given some general presentations about it during its various stages of development, some of which are online [2]. PipeWire started as a way to share arbirary multimedia, wich requires vastly different requirements regarding format support, device and memory management than JACK. It wasn't until I started experimenting with audio processing that the design started to gravitate to JACK. And then some of JACKs features became a requirement for PipeWire. The end goal of PipeWire is to interconnect applications and devices through a shared graph in a secure and efficient way. Some of the first applications will be wayland screen sharing and camera sharing with access control for sandboxed applications. It would be great if we could also use this to connect audio apps and devices, possibly unifying the pulseaudio/JACK audio stack. Because the general design is, what I think, now very similar to JACK, many people have been asking me if I'm collaborating with the linux pro-audio community on this in any way at all. I have not but I really want to change that. In this mail I hope to start a conversation about what I'm doing and I hope to get some help and experience from the broader professional audio developers community on how we can make this into something useful for everybody. I've been looking hard at all the things that are out there, including Wayland, JACK, LV2, CRAS, GStreamer, MFT, OMX,.. and have been trying to combine the best ideas of these projects into PipeWire. A new plugin API was designed for hard realtime processing of any media type. PipeWire is LGPL licensed and depends only on a standard c library. It's currently targeting Linux. At the core of the PipeWire design is a graph of processing nodes with arbirary input/output ports. Before processing begins, ports need to be configured with a format and a set of buffers for the data. Buffer data and metadata generally lives in memfd shared memory but can also be dmabuf or anything that can be passed as an fd between processes. There is a lot of flexibility in doing this setup, reusing much of the GStreamer experience there is. This all happens on the main thread, infrequently, not very important for the actual execution of the graph. In the realtime thread (PipeWire currently has 1 main thread and 1 realtime data thread), events from various sources can start push/pull operations in the graph. For the purpose of this mail, the audio sink uses a timerfd to wake up when the alsa buffer fill level is below a threshold. This causes the sink to fetch a buffer from its input port queue and copy it to the alsa ringbuffer. It then issues a pull to fetch more data from all linked peer nodes for which there is nothing queued. These peers will then eventually push another buffer in the sink queue to be picked up in the next pull cycle of the sink. This is somewhat similar to the JACK async scheduling model. In the generic case, PipeWire has to walk upstream in the graph until it finds a node that can produce something (see below how this can be optimized). Scheduling of nodes is, contrary to JACKs (and LADSPA and LV2) single 'process' method, done with 2 methods: process_input and process_ouput. This is done to support more complex plugins that need to decouple input from output and to also support a pull model for plugins. For internal clients, we directly call the methods, for external clients we use an eventfd and a shared ringbuffer to send the right process command to the client. When the external client has finished processing or need to pull, it signals PipeWire, which then wakes up the next clients if needed. This is different from JACK, where a client directly wakes up the peers to avoid a server context switch. JACK can do this because the graph and all client semaphores are shared. PipeWire can't in general for a couple of reaons: 1) you need to bring mixing of arbitrary formats to the clients 2) sandboxed clients should not be trusted with this information and responsability. In some cases it would probably be possible to improve that in the future (see below). This kind of scheduling works well for generic desktop style audio and video. Apps can send buffers of the size of their liking. Bigger buffers means higher latency but less frequent wakeups. The sink wakeup frequency is determined by the smallest buffer size that needs to be mixed. There is an upper limit for the largest amount of data that is mixed in one go to avoid having to do rewinds in alsa and still have reasonable latency when doing volume changes or adding new streams etc. The idea is to make a separate part of the graph dedicated to pro-audio. This part of the graph runs with mono 32bit float sample buffers of a fixed size and samplerate. The nodes running in this part of the graph also need to have a fixed input-output pattern. In this part of the graph, negotiating the format becomes trivial. We can preallocate a fixed size buffer for each port that is used to send/mix data between nodes. Exactly like how JACK works. In this scenario it would be possible to bring some of the graph state to trusted clients so that they can wake up their peers directly. As it turns out, the generic scheduling mechanism simplifies to the JACK way of scheduling and the option to do some optimisations (can directly start push from the sources, bundle process_input/output calls, mixing on ports is simplified by equal buffer sizes, ...) There is a lot more stuff that I can talk about and a lot of things that need to be fleshed out like latency calculations, an equivalent of JACK transport, session management, ... But this mail is already getting long :) I would very much like to hear your ideas, comments, flames, thoughts on this idea. I think I'm at a stage where I can present this to a bigger audience and have enough experience with the matter to have meaningful discussions. PipeWire is currently still in heavy development, many things can and do still change. I'm currently writing a replacement libjack.so[3] that runs jack clients directly on PipeWire (mixing and complicated scheduling doesn't work yet). Hope to hear your comments, Wim Taymans [1] pipewire.org [2] https://www.youtube.com/watch?v=6Xgx7cRoS0M [3] https://github.com/PipeWire/pipewire-jack

7 years, 2 months

revisiting system timer sound application wake up

by Ricardo Biehl Pasquali

08/08/2018 (ignore this date) PulseAudio tsched ================= Around December 2007 to February 2009 Lennart Poettering has asked some questions in alsa-devel mailing list in order to implement the timer based audio scheduling for PulseAudio. In 2008 he wrote an article named "What's Cooking in PulseAudio's glitch-free Branch". I have not found detailed documentation about PulseAudio timer wake up and I'm not sure about some implementation details. By using the emails from Lennart and his article a Documentation can start to grow. Do PulseAudio synchronize to sound device by adjusting the timer or the amount committed (application pointer)? The nice thing about the second case is that the period time (in application's view) is bonded to system clock instead of sound device's. What timer interface does PulseAudio use? timerfd? POSIX timers? Hardware issues =============== How the deviation of system and sound device clock is today? Perhaps someone that works with sound cards or SoCs can answer this. Do the most of today's hardware has the same clock for sound device and system? How sound devices deal with random accesses to their ring buffer. Do most of them allow writing/reading anywhere and anytime? How is the acurracy and precision of the hardware pointer returned by sound devices? Does SNDRV_PCM_INFO_BATCH flag reveals something about the above one? As PulseAudio runs in many Linux distributions today, it seems an interesting idea to implement an utility to collect some information regarding the above issues. Then make it available to the public. E.g. it would be possible to know the pointer acurracy for specific devices. SCHED_DEADLINE and timerfd ========================== There are a few works over there about SCHED_DEADLINE and audio: - This paper presented in Linux Audio Conference 2011: BAGNOLI, Giacomo; CUCINOTTA, Tommaso; FAGGIOLI, Dario. 2011. Low-Latency Audio on Linux by Means of Real-Time Scheduling. Dario Faggioli is one of the authors of SCHED_DEADLINE. - This presentation of Alessio Balsini: "Experimenting with the Android Audio Pipeline and SCHED_DEADLINE". OSPM 2018. Here is an overview: <https://lwn.net/Articles/754923/>. In SCHED_DEADLINE I assume there is no way to accurately adjust the timer. As well as there is no way to adjust when the first expiration will be. I think this is because of Constant Bandwidth Server algorithm. Must ask to some developer of this. Having this in mind, the only way to synchronize to sound device is by adjusting the amount read/written (application pointer). Timerfd on the other hand is fully adjustable. However, it can be preempted by a high priority task. Anyway I don't see that as a problem as it should ralely happen. Perhaps SCHED_DEADLINE is suitable for professional environment (aiming almost-zero glitches). Thoughts on this? Is there someone planning to implement SCHED_DEADLINE in PulseAudio? I'm currently working on both implementations (not for PulseAudio). See <https://github.com/ricardobiehl/simplesound>. Take a look at 'Documentation/timer_wakeup.rst', 'tools/timer_wakeup.c' and 'tools/deadline_wakeup.c'. 'tools/waveplay.c' play .wav files using the application wakeup method specified in 'tools/Makefile'. Documentation ============= Many of the points I've mentioned should get well covered by some documentation (e.g. ALSA, PulseAudio). Also, I think ALSA's internal (kernel-level) documentation needs to grow. I'm actually working on that, documenting some stuff as I'm developing a small sound library that interacts directly to ALSA in kernel, and the application wakeup using timerfd and SCHED_DEADLINE. Cheers!

7 years, 2 months

All videos online, website going to read-only mode

by Linux Audio Conference

Dear Linux Audio community, we're sending this mail to let you know about the availability of the remaining videos from LAC2018. You can find them on media.ccc.de [1] and on the dedicated event pages linked to in the schedule [2]. We hope you had a great time at the conference and if you couldn't be there physically, this is now the time to have a look at much of what has happened in Berlin this year. In other news, the website [3] is going to read-only mode shortly. See you at future LACs! [1] https://media.ccc.de/b/conferences/lac/lac18 [2] https://lac.linuxaudio.org/2018/pages/schedule/ [3] https://lac.linuxaudio.org/2018/ -- Linux Audio Conference team

7 years, 2 months

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