Linux-audio-dev February 2023

linux-audio-dev@lists.linuxaudio.org

18 participants
12 discussions

[LAD] stereospread.lv2 - Stereo spreading plugin (conjugate random phase all-pass filter pair)

by Florian Paul Schmidt

Hi, I wrote a small plugin for spreading mono content to stereo and I would like to receive some feedback on it from people more knowledgable in DSP matters than me. It has one nice property: It spreads the signal over the spectrum by way of a conjugate pair of random phase all-pass filters, and since it's a conjugate pair it sums back to unity when down-mixing to mono. So, no ugly comb filter effects like when downmixing a Haas-expanded signal. https://github.com/fps/stereospread.lv2 The page has an example sound.. How it works: It's actually rather simple: 1. create a vector of random phases (matlab notation): filter_length = 1000; spread = pi; hfft1 = exp(-1i*(spread*rand(filter_length,1) - spread/2)); And for the second filter just take the complex conjugate: hfft2 = conj(hfft1); This ensures that what is a phase theta in the first filter becomes a phase of -theta in the second filter, and summed that just gives a phase of 0. 2. Then assemble the coefficients such that it corresponds to a fft of a real signal and do the inverse fft (possibly I have a small error here which i needed to fix with the 'symmetric' flag in matlab): f1 = ifft([1; hfft1; conj(hfft1((end-1):-1:1))], 'symmetric'); f2 = ifft([1; hfft2; conj(hfft2((end-1):-1:1))], 'symmetric'); 3. The two IRs f1 and f2 implement the pair of filters and can be applied via convolution (which the above plugin does). It seems to be possible with little ill effect to reduce the length of the filter down to a size of 50 samples or so by just cutting it off before the convolution. What do you think? Kind regards, FPS -- https://dfdx.eu

2 years, 8 months

[LAD] I'm being stalked, please be weary

by Jeanette C.

Hey hey, I am being stalked and the stalker has already contacted people I am connected with. Please, if anyone contacts you with any request, be weary. It's a plain name gmail address. Thank you and sorry to bother you! Best wishes, Jeanette -- * Website: http://juliencoder.de - for summer is a state of sound * Youtube: https://www.youtube.com/channel/UCMS4rfGrTwz8W7jhC1Jnv7g * Audiobombs: https://www.audiobombs.com/users/jeanette_c * GitHub: https://github.com/jeanette-c Do you even know what I like Just what I'm living for, What I adore <3 (Britney Spears)

2 years, 9 months

[LAD] Status of Pipewire

by Fons Adriaensen

Hello all, I've been contemplating trying out Pipewire as a replacement for Jack. What is holding me back is a what seems to be a serious lack of information. I'm not prepared to spend a lot of time and risk breaking a perfectly working system just to find out that it was a bad idea from the start. So I have a lot of questions which maybe some of you reading this can answer. Thanks in advance for all useful information. A first thing to consider is that I actually *like* the separation of the 'desktop' and 'pro audio' worlds that using Jack provides. I don't want the former to interfere (or just be able to do so) with the latter. Even so, it may be useful in some cases to route e.g. browser audio or a video conference to the Jack world. So the ideal solution for me would be the have Pipewire as a Jack client. So first question: Q1. Is that still possible ? If not, why not ? If the answer is no, then all of the following become relevant. Q2. Does Pipewire as a Jack replacement work, in a reliable way [1], in real-life conditions, with tens of clients, each maybe having up to a hundred ports ? Q3. What overhead (memory, CPU) is incurred for such large systems, compared to plain old Jack ? A key feature of Jack is that all clients share a common idea of what a 'period' is, including its timing. In particular the information provided by jack_get_cycle_times(), which is basically the state of the DLL and identical for all clients in any particular period. Now if Pipewire allows (non-Jack) clients with arbitrary periods (and even sample rates) Q4. Where is the DLL and what does it lock to when Pipewire is emulating Jack ? Q5. Do all Jack clients see the same (and correct) info regarding the state of the DLL in all cases ? The only way I can see this being OK would be that the Jack emulation is not just a collection of Pipewire clients which happen to have compatible parameters, but actually a dedicated subsystem that operates almost independently of what the rest of Pipewire is up to. Which in turn means that having Pipewire as a Jack client would be the simpler (and hence preferred) solution. [1] which means I won't fall flat on my face in front of a customer or a concert audience because of some software hickup. Ciao, -- FA

2 years, 10 months

[LAD] SoundTracker 1.0.4-pre1 released

by Yury Alyaev

Hi all, I just have released SoundTracker v1.0.4-pre1. I've decided to make a feature-freeze at this point, this means that 1.0.4 will have no new features comparing to this pre-release, only fixes and small improvements if something will be found inconvenient. So I ask everyone who has interest in SoundTracker to test this release. Here are main features of the 1.0.4-pre1 release comparing to 1.0.3: * General: - Full-screen mode - Volume of all samples of an instrument / all samples in the module can be set to a value, not only gained. Also all samples panning can be adjusted. These functions can also modify the explicitly given values of volume / panning of notes in patterns - Improved compatibility with FastTracker, also MOD files are played more correctly - Unused patterns are listed in the Pattern Sequence Editor - Sampling rate can be specified while saving a sample as WAV - New module optimizer with many control parameters * Track Editor: - Moving notes up / down is implemented * Sample Editor: - External programs can be used for samples processing. The interaction between such a program is described using XML spec, no ST recompiling required - The whole sample is drawn after recording - Exponential / reverse exponential volume fade transients * Instrument Editor: - Envelope inversion and shifting is implemented SoundTracker download page: https://sourceforge.net/projects/soundtracker/files/ Regards, Yury.

2 years, 11 months

[LAD] Timed MIDI event handling

by Jeanette C.

Hey hey, I want to record MIDI into a sequencer application, using RtMidi. My understanding is: inside that application's tracks events are directly stamped with delta times, measured in ticks, since the last event in that track. Internally the sequencer runs a clock at a high rate (480PPQ or higher). It's probably best to "quantise" events to that internal clock while they are recorded. So the track object, with its connected MIDI input device, must be linked to the clock, which will wake up, whenever it is time for a tick. Linked meaning there must be some way to exchange relevant info (one of them owning a reference, callback, shared data, ...) If that basic, and simplified, principle is sound. Which solutions have been proven feasible to achieve that? For example: make every active track a thread and have a global mutex? Each track registers a callback function with the clock to store events. ...? RtMidi's input object (RtMidiIn) offers to register a callback function, that will be called with a received MIDI message, when input is available. Such messages could be stored in a container and be delivered when it's tick time. -- So again, a track might own a reference to the Clock thread object and query the current tick count since the last record start command and use that information to calculate the correct delta. Not sure though, whether that might raise issues with execution times. Using an atomic variable as a tick counter will avoid having to use mutex locks, at least. Is there any practical advice? Something based on experience? A good read? Best wishes and thanks, Jeanette -- * Website: http://juliencoder.de - for summer is a state of sound * Youtube: https://www.youtube.com/channel/UCMS4rfGrTwz8W7jhC1Jnv7g * Audiobombs: https://www.audiobombs.com/users/jeanette_c * GitHub: https://github.com/jeanette-c But now I'm stronger than yesterday <3 (Britney Spears)

2 years, 11 months

[LAD] Calculating logarithmic curve for controller automation

by Jeanette C.

Hey hey, how is a logarithmic curve usually programmed in a DAW or sequencer? Do you scale the values of (log1) to log(2) to the desired range and stretch it over time? Do you ajudst steepness by either using more less of the log function or changing both values like log(20) to log(21)? I'm sure there are many ways to do it, but I'd assume that there are a few "classic" ways to approach this with anything from controller "automation" to envelope curve settings to ... I'd appreciate any hints or simple examples from software practise. Best wishes and thanks, Jeanette -- * Website: http://juliencoder.de - for summer is a state of sound * Youtube: https://www.youtube.com/channel/UCMS4rfGrTwz8W7jhC1Jnv7g * Audiobombs: https://www.audiobombs.com/users/jeanette_c * GitHub: https://github.com/jeanette-c Baby, take the time to realize I'm not the kind to sacrifice <3 (Britney Spears)

2 years, 11 months

[LAD] Tempo change functions.

by Fons Adriaensen

Hello all, What exactly is meant by a linear or exponential tempo change ? Is the tempo a lin/exp function of time, or of score position ? A bit of algebra leads to this: Let t = time (seconds) p = score position (beats) v = tempo (beats / second) [1] We have v = dp / dt # by definition If v is a linear function of t, then v (p) = square root of a linear fuction of p If v is a exponential function of t, then v (p) = a linear function of p if v is a linear function of p then v (t) = an exponential function of t If v is an exponential function of p, then v (t) = inverse of a linear function of t So there is plenty of room for confusion... [1] Using SI units :-) Ciao, -- FA

2 years, 11 months

[LAD] Admissable clock inaccuracy for MIDI

by Jeanette C.

Hey hey, I'm experimenting with c++ trying to program something nice for MIDI. I'm now experimenting with clocks, using both the standard c++ chrono library and the boost chrono library. My example program sets a desired delta between ticks of 250ms. I see that there is a difference, since Boost chrono can also use pthread thread parameters foor realtime scheduling and priorities. With the standard chrono and thread library my ticks are usually out by anything between 150.000 to 290.000 nanoseconds. Using boost a tick is out anything between 110.000 to 124.000 nanoseconds. Yes, much better. But, assuming that I correct the drift, does it make a difference for MIDI instruments? I know one of my MIDI instruments with a clock sync'able delay can be rather touchy with MIDI clocks, but are there good estimated guidelines experience values how precise ticks should be? Best wishes and thanks for any help, Jeanette -- * Website: http://juliencoder.de - for summer is a state of sound * Youtube: https://www.youtube.com/channel/UCMS4rfGrTwz8W7jhC1Jnv7g * Audiobombs: https://www.audiobombs.com/users/jeanette_c * GitHub: https://github.com/jeanette-c Top down, on the strip Lookin' in the mirror I'm checkin' out my lipstick <3 (Britney Spears)

2 years, 11 months

[LAD] GRAME and Faust selected for Google Summer of Code 2023

by Stéphane Letz

Hi All, GRAME and Faust are selected for Google Summer of Code 2023: https://summerofcode.withgoogle.com/programs/2023/organizations/grame Projets ideas here: https://github.com/grame-cncm/faustideas Contributions welcome ! Stéphane

2 years, 11 months

[LAD] update (again)

by Fons Adriaensen

Hello all, Just hours after the release of zita-resampler-1.10.0 yesterday, a new bug was reported, not in the library but in the zresample application. This is now fixed in 1.10.1 Ciao, -- FA

2 years, 11 months

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Linux-audio-dev February 2023