Linux-audio-dev

linux-audio-dev@lists.linuxaudio.org

4 participants
7502 discussions

by John Rigg

Good to know there's a new hdspm driver being worked on. > The settings of the card can nevertheless be graphically > set, using alsamixer in the terminal. I've only had limited success using alsamixer to change settings on HDSPe MADI cards. I've found amixer and alsactl (both CLI) more useful here. I made a web page describing their use with the HDSPe MADI: http://www.jrigg.co.uk/linuxaudio/hdspe-madi.html Much of it should also be applicable to the HDSPe AES32 (allowing for different parameter names and channel numbers). John

3 years, 9 months

status of RME HDSPe AES (32 channels)

by lievenmoors＠protonmail.com

Hi, I've been searching all day through the linus audio mailing list archives, and I couldn't find any confirmation that this card is working on Linux. When I look at the source code of the hdspm driver, I see lot's of references to the AES32, so I suppose it is at least partly supported. Is there anyone here that knows more about this card? Thanks! lieven

3 years, 9 months

[ANN] Vee One Suite 0.9.13 - A QStuff* Spring'20 Release batch #2

by Rui Nuno Capela

Greetings once more, The 'Vee One Suite' of so called 'old-school' software instruments, synthv1 [1] as a polyphonic subtractive synthesizer, samplv1 [2] a polyphonic sampler synthesizer, drumkv1 [3] as yet another drum-kit sampler and padthv1 [4] as a polyphonic additive synthesizer, are all being released now for the second 'QStuff*' Spring'20 batch of the (quarantine) season. All still available in dual form deliveries: - a pure stand-alone JACK client with JACK-session [5], NSM (Non Session management [6]) and both JACK MIDI and ALSA MIDI [7] input support; - a LV2 instrument plug-in [8]. Not much of a ravishing change-log, just on the brink of utterly boring maintenance stuff: - Always keep the current sample loop-cycle on note-off, while entering a DCA Release stage. (applies to samplv1 [2] only) - Maximum pitch-bend range has been expanded to 0..400%, allegedly to support some PME driven instruments. - Make man page compression reproducible (after request by Jelle van der Waa, thanks). - Fixed crash when showing the tooltip of a negative note number that results from dragging over and beyond the left edge of the virtual piano keyboard. - Remove -v (verbose) flag from 'strip' post-link command. - Fixed CMake build by adding missing Custom Color Theme (palette) form file (.ui). - Bumped copyright headers into the New Year (2020). The 'Vee One Suite' are free, open-source Linux Audio [10] software, distributed under the terms of the GNU General Public License (GPL) [11] version 2 or later. **synthv1 - an old-school polyphonic synthesizer [1] ** synthv1 0.9.13 (spring'20) released! synthv1 is an old-school all-digital 4-oscillator subtractive polyphonic synthesizer with stereo fx. LV2 URI: http://synthv1.sourceforge.net/lv2 website: https://synthv1.sourceforge.io http://synthv1.sourceforge.net project page: https://sourceforge.net/projects/synthv1 downloads: https://sourceforge.net/projects/synthv1/files - source tarball: https://download.sf.net/synthv1/synthv1-0.9.13.tar.gz - source package: https://download.sf.net/synthv1/synthv1-0.9.13-53.rncbc.suse.src.rpm - binary packages (openSUSE Tumbleweed): https://download.sf.net/synthv1/synthv1-0.9.13-53.rncbc.suse.x86_84.rpm https://download.sf.net/synthv1/synthv1-lv2-0.9.13-53.rncbc.suse.x86_84.rpm - AppImage [9] package (JACK stand-alone only): https://download.sf.net/synthv1/synthv1-0.9.13-53.x86_64.AppImage git repos: https://git.code.sf.net/p/synthv1/code https://github.com/rncbc/synthv1.git https://gitlab.com/rncbc/synthv1.git https://bitbucket.org/rncbc/synthv1.git **samplv1 - an old-school polyphonic sampler [2]** samplv1 0.9.13 (spring'20) released! samplv1 is an old-school polyphonic sampler synthesizer with stereo fx. LV2 URI: http://samplv1.sourceforge.net/lv2 website: https://samplv1.sourceforge.io http://samplv1.sourceforge.net project page: https://sourceforge.net/projects/samplv1 downloads: https://sourceforge.net/projects/samplv1/files - source tarball: https://download.sf.net/samplv1/samplv1-0.9.13.tar.gz - source package: https://download.sf.net/samplv1/samplv1-0.9.13-53.rncbc.suse.src.rpm - binary package (openSUSE Tumbleweed): https://download.sf.net/samplv1/samplv1-0.9.13-53.rncbc.suse.x86_84.rpm https://download.sf.net/samplv1/samplv1-lv2-0.9.13-53.rncbc.suse.x86_84.rpm - AppImage [9] package (JACK stand-alone only): https://download.sf.net/samplv1/samplv1-0.9.13-53.x86_64.AppImage git repos: https://git.code.sf.net/p/samplv1/code https://github.com/rncbc/samplv1.git https://gitlab.com/rncbc/samplv1.git https://bitbucket.org/rncbc/samplv1.git **drumkv1 - an old-school drum-kit sampler [3]** drumkv1 0.9.13 (spring'20) released! drumkv1 is an old-school drum-kit sampler synthesizer with stereo fx. LV2 URI: http://drumkv1.sourceforge.net/lv2 website: https://drumkv1.sourceforge.io http://drumkv1.sourceforge.net project page: https://sourceforge.net/projects/samplv1 downloads: https://sourceforge.net/projects/drumkv1/files - source tarball: https://download.sf.net/drumkv1/drumkv1-0.9.13.tar.gz - source package: https://download.sf.net/drumkv1/drumkv1-0.9.13-43.rncbc.suse.src.rpm - binary package (openSUSE Tumbleweed): https://download.sf.net/drumkv1/drumkv1-0.9.13-43.rncbc.suse.x86_84.rpm https://download.sf.net/drumkv1/drumkv1-lv2-0.9.13-43.rncbc.suse.x86_84.rpm - AppImage [9] package (JACK stand-alone only): https://download.sf.net/drumkv1/drumkv1-0.9.13-53.x86_64.AppImage git repos: https://git.code.sf.net/p/drumkv1/code https://github.com/rncbc/drumkv1.git https://gitlab.com/rncbc/drumkv1.git https://bitbucket.org/rncbc/drumkv1.git **padthv1 - an old-school polyphonic additive synthesizer [4]** padthv1 0.9.13 (spring'20) released! padthv1 is an old-school polyphonic additive synthesizer with stereo fx. padthv1 is based on the PADsynth algorithm by Paul Nasca, as a special variant of additive synthesis. LV2 URI: http://padthv1.sourceforge.net/lv2 website: http://padthv1.sourceforge.net https://padthv1.sourceforge.io project page: https://sourceforge.net/projects/padthv1 downloads: https://sourceforge.net/projects/padthv1/files - source tarball: https://download.sf.net/padthv1/padthv1-0.9.13.tar.gz - source package: https://download.sf.net/padthv1/padthv1-0.9.13-53.rncbc.suse.src.rpm - binary package (openSUSE Tumbleweed): https://download.sf.net/padthv1/padthv1-0.9.13-53.rncbc.suse.x86_84.rpm https://download.sf.net/padthv1/padthv1-lv2-0.9.13-53.rncbc.suse.x86_84.rpm - AppImage [9] package (JACK stand-alone only): https://download.sf.net/padthv1/padthv1-0.9.13-53.x86_64.AppImage git repos: https://git.code.sf.net/p/padthv1/code https://github.com/rncbc/padthv1.git https://gitlab.com/rncbc/padthv1.git https://bitbucket.org/rncbc/padthv1.git References: [1] synthv1 - an old-school polyphonic synthesizer https://synthv1.sourceforge.io http://synthv1.sourceforge.net/ [2] samplv1 - an old-school polyphonic sampler https://samplv1.sourceforge.io http://samplv1.sourceforge.net/ [3] drumkv1 - an old-school drum-kit sampler https://drumkv1.sourceforge.io http://drumkv1.sourceforge.net/ [4] padthv1 - an old-school polyphonic additive synthesizer https://padthv1.sourceforge.io http://padthv1.sourceforge.net/ [5] JACK Audio Connection Kit https://jackaudio.org/ [6] NSM, Non Session Management http://non.tuxfamily.org/nsm/ [7] ALSA, Advanced Linux Sound Architecture https://www.alsa-project.org/ [8] LV2, Audio Plugin Standard, the extensible successor of LADSPA http://lv2plug.in/ [9] AppImage, Linux apps that run anywhere https://appimage.org/ [10] Linux Audio consortium of libre software for audio-related work https://linuxaudio.org [11] GNU General Public License https://www.gnu.org/copyleft/gpl.html See also: https://www.rncbc.org/drupal/node/2076 Enjoy && Stay safe! -- rncbc aka. Rui Nuno Capela rncbc(a)rncbc.org

3 years, 9 months

Mackie Onyx 1640i: Long run playback results in xrun and hang.

by AreYouLoco?

Hi, I am using custom build real-time kernel 5.12.2-rt3-fw. JACK with ALSA backend on Mackie Onyx 1640i. Lenovo Thinkpad T420 with FireWire. Debian Bullseye (11.0 - stable) JACKD version 1.9.17. I am using 512 samples 3 periods @48kHz. The problem is that long run playback only around 1h (but the time differs), result in XRUN detected by JACK logs and the whole audio stops. So I have to restart JACK in order to make it work. DMESG is silent about that issue. But sometimes I get: snd_oxfw fw1.0: Detect discontinuity of CIP: E8 00 And different hex values there. Here is interesting part from the log: Sun Sep 12 22:10:12 2021: Jack: JackRequest::Notification Sun Sep 12 22:10:12 2021: Jack: **** alsa_pcm: xrun of at least 0.027 msecs Sun Sep 12 22:10:12 2021: Jack: Repreparing capture Sun Sep 12 22:10:12 2021: Jack: Repreparing playback Sun Sep 12 22:10:12 2021: Jack: ALSA XRun wait_status = 0 Sun Sep 12 22:10:12 2021: Jack: JackEngine::ClientNotify: no callback for notification = 3 Sun Sep 12 22:10:12 2021: Jack: JackEngine::ClientNotify: no callback for notification = 3 Sun Sep 12 22:10:12 2021: Jack: JackClient::ClientNotify ref = 2 name = dbusapi notify = 3 Sun Sep 12 22:10:12 2021: Jack: JackClient::kXRunCallback Sun Sep 12 22:10:12 2021: Jack: JackEngine::ClientNotify: no callback for notification = 3 Sun Sep 12 22:10:12 2021: Jack: JackEngine::ClientNotify: no callback for notification = 3 Sun Sep 12 22:10:12 2021: Jack: JackEngine::ClientNotify: no callback for notification = 3 Sun Sep 12 22:10:12 2021: Jack: JackEngine::ClientNotify: no callback for notification = 3 Sun Sep 12 22:10:28 2021: ERROR: ALSA: poll time out, polled for 15998655 usecs, Retrying with a recovery, retry cnt = 1 Sun Sep 12 22:10:44 2021: ERROR: ALSA: poll time out, polled for 15998654 usecs, Retrying with a recovery, retry cnt = 2 Sun Sep 12 22:11:00 2021: ERROR: ALSA: poll time out, polled for 15998659 usecs, Retrying with a recovery, retry cnt = 3 Sun Sep 12 22:11:16 2021: ERROR: ALSA: poll time out, polled for 15998649 usecs, Retrying with a recovery, retry cnt = 4 Anyone have idea what might be the cause of this XRUN? Or how to debug that? Its reproducible on my setup. Another log catched the same issue: Sun Sep 12 16:50:44 2021: Jack: **** alsa_pcm: xrun of at least 0.030 msecs Sun Sep 12 16:50:44 2021: Jack: Repreparing capture Sun Sep 12 16:50:44 2021: Jack: Repreparing playback Sun Sep 12 16:50:44 2021: Jack: ALSA XRun wait_status = 0 Sun Sep 12 16:50:44 2021: Jack: JackRequest::Notification Sun Sep 12 16:50:44 2021: Jack: JackEngine::ClientNotify: no callback for notification = 3 Sun Sep 12 16:50:44 2021: Jack: JackEngine::ClientNotify: no callback for notification = 3 Sun Sep 12 16:50:44 2021: Jack: JackClient::ClientNotify ref = 2 name = dbusapi notify = 3 Sun Sep 12 16:50:44 2021: Jack: JackClient::kXRunCallback Sun Sep 12 16:50:44 2021: Jack: JackEngine::ClientNotify: no callback for notification = 3 Sun Sep 12 16:50:44 2021: Jack: JackEngine::ClientNotify: no callback for notification = 3 Sun Sep 12 16:50:44 2021: Jack: JackEngine::ClientNotify: no callback for notification = 3 Sun Sep 12 16:50:44 2021: Jack: JackEngine::ClientNotify: no callback for notification = 3 Sun Sep 12 16:51:00 2021: ERROR: ALSA: poll time out, polled for 15998366 usecs, Retrying with a recovery, retry cnt = 1 Sun Sep 12 16:51:16 2021: ERROR: ALSA: poll time out, polled for 15998353 usecs, Retrying with a recovery, retry cnt = 2 Sun Sep 12 16:51:32 2021: ERROR: ALSA: poll time out, polled for 15998352 usecs, Retrying with a recovery, retry cnt = 3 Sun Sep 12 16:51:48 2021: ERROR: ALSA: poll time out, polled for 15998367 usecs, Retrying with a recovery, retry cnt = 4 Sun Sep 12 16:52:04 2021: ERROR: ALSA: poll time out, polled for 15998353 usecs, Retrying with a recovery, retry cnt = 5 Sun Sep 12 16:52:20 2021: ERROR: ALSA: poll time out, polled for 15998371 usecs, Reached max retry cnt = 5, Exiting Sun Sep 12 16:52:20 2021: ERROR: JackAudioDriver::ProcessAsync: read error, stopping... Sun Sep 12 16:52:20 2021: Jack: JackPosixThread::ThreadHandler : exit I am willing to provide additional info. Thank you in advance, AreYouLoco?

3 years, 9 months

B.Angr 1.6.0

by Sven Jaehnichen

Hi, a new release of B.Angr is out now. B.Angr is a multi-dimensional dynamically distorted staggered multi-bandpass LV2 plugin. B.Angr basically consists of four parameter instances of Airwindows XRegion. XRegion itself is a distorted multi-bandpass filter plugin. B.Angr fades automatically or manually between the four parameter instances. The result is a dynamic and heavily distorted sound. See the preview video. What's new: * Faster * Optimized DSP code * Compiler optimizations (`-O3 -ffast-math`) * Support lv2:Parameters, lv2:State * Cursor XY can now externally be controlled * Binary compatibility improved * Static libs * Strip symbols by default * Binary packages provided Github: https://github.com/sjaehn/BAngr/ Releases: https://github.com/sjaehn/BAngr/releases/tag/1.6.0 Preview video: https://www.youtube.com/watch?v=-kWy_1UYazo <https://www.youtube.com/watch?v=-kWy_1UYazo> Regards, Sven

3 years, 9 months

Phase rotation

by Robin Gareus

Hi all, During the last days, I looked into phase-ration: components of a signal are delayed differently depending on their frequency. A good write up on the subject can be found at [1], and a commercial tool is available from [2]. The interesting aspect is that phase rotation does not alter the sound of the signal nor the loudness. However changing the phase vs. frequency relationship between lower and upper harmonics changes the waveform and can affect where the digital peak occurs. For this reason phase rotation is commonly used by radio stations to reduce the signal peak, and make the signal more symmetrical. Phase rotation circuits are also used during mastering to increase headroom. To implementing this, the following operations are performed: * FFT * rotate phase, retain magnitude * inverse FFT This works well, except for the first FFT bin: 0 Hz, DC offset. If the phase-shift changes the average DC level of the signal there is a discontinuity. I could use some of the collective expertise of this list. Has anyone looked into this before? Is it even possible with piecewise block processing? Is there a way to calculate the DC offset? Since the overall magnitude does not change simply summing up the bins results in the same 0th bin. I considered to low-pass filter the DC offset, or simply remove the DC offset using a high-pass before the FFT to mitigate the issue. But either introduces different artifacts. If I understand correctly, a Hilbert transform as mentioned at [1] has the same issue. I've condensed the DSP into a simple test tool to toy around with the issue: https://gist.github.com/17fd61b04d4c4939727dfdccd79f53a5 with low frequencies, the differences are obvious: * https://i2.paste.pics/cc32f6c5d622e31ab0e830ab1ce205e9.png (2.5 * FFT's base-freq) * https://i2.paste.pics/3e6eb827148b90800569843c11da2e48.png (0.37 * FFT's base-freq) I'd appreciate any insights. -- robin [1] https://forum.juce.com/t/how-to-rotate-the-phase-of-an-audio-signal/39072/10 [2] https://www.izotope.com/en/products/rx/features/phase.html

3 years, 10 months

MIDI program change

by Will Godfrey

I've been trying to get hold of information on MIDI-2. So far I've only got overviews and limited data format info. But one thing I saw today was definitely a YES! moment. For years, I've been battling against DAWs and sequencers that insist on generating bank messages when none were sent, and then almost always set it to Bank MSB=LSB=0. The MIDI-1 spec was never 100% clear on that, but the MIDI-2 one says exactly what I've maintained all along. When sending a program change, bank change is optional! It is perfectly valid to want to change instrument on the same bank, and indeed all the hardware synths I've ever owned work in this way. -- Will J Godfrey https://willgodfrey.bandcamp.com/ http://yoshimi.github.io Say you have a poem and I have a tune. Exchange them and we can both have a poem, a tune, and a song.

3 years, 10 months

Changes in ALSA firewire stack at Linux kernel v5.14 release

by Takashi Sakamoto

Dear all, I'm a maintainer of ALSA firewire stack. Linux kernel v5.14 was out a few days ago[1], including some changes in ALSA firewire stack. The changes bring improvement for usage of including drivers by solving some issues. I appreciate the users cooperating for it[2]. This message includes two topics about solved issues in the release: 1. get rid of playback noise by recovering media clock 2. allow some applications to run without periodical hardware interrupts and another topic: 3. device aggregation Let me describe the two topics first. 1. get rid of playback noise by recovering media clock Many users had been reporting playback noise since the initial version of each driver in ALSA firewire stack. The cause of the issue is complicated to explain, but let me roughly summarize it to a point below: * mismatch between audio sample count in playback stream and the one expected by hardware Since the initial stage of ALSA firewire stack, included drivers transfer audio data frames per second the exact count as sampling frequency, which is configured via ALSA PCM interface; e.g. 44.1 kHz. But it is figured out that it is not suitable for many models. For recent years, I've measured actual packets from/to various models with Windows and OS X drivers[3], and realized the below phenomena. Here, the configured frequency is called 'nominal', and the measured frequency is called 'effective'. * the effective frequency is not the same as the nominal frequency, less or greater by several audio data frames (<= 10 frames) * the effective frequency is not even in successive seconds for some models The phenomena mean that it is not achieved to transfer samples for playback sound by nominal frequency, and computation for even number of samples per second for some models. Additionally, in isochronous communication of IEEE 1394, part of models support time stamp per isochronous packet[4]. When parsing the sequence of time stamp and compare it to frequency of samples included in the packets, I realize the phenomena below: * the phase of sample based on computed time stamp shifts during long packet streaming * before and after configuring source of sampling clock to external signal input such as S/PDIF, neither the effective frequency of samples in packets nor the sequence of time stamp in packets have difference. The phenomena give us some insights. At least, the phase of samples is not deterministic somehow in driver side. It is required to recover the timing to put audio data frame into packet according to packets transferred by the hardware. This is called 'media clock recovery'[5]. In engineering field, many method of media clock recovery has been invented for each type of media. The way which ALSA firewire stack in v5.14 uses is the simplest one. It is to replay the sequence in transferred packets[6][7][8]. The result looks better. As long as I tested, it can cover all of models supported by ALSA firewire stack. 2. allow applications to run independently of periodical hardware interrupts ALSA PCM interface has hardware parameter for runtime of PCM substream to process audio data frame without scheduling periodical hardware interrupts[9]. PulseAudio and PipeWire use the function. All of drivers[10] in ALSA firewire stack now support it. Linux FireWire subsystem has function to flush queued packet till the most recent isochronous cycle. The function is available in process context without waiting for scheduled hardware interrupts, and allows drivers to achieve the topic. In most cases, it's done by calling ioctl(2) with SNDRV_PCM_IOCTL_HWSYNC. The call is the part of routine in usual ALSA PCM application, thus users do not need to take extra care of it. I think these improvements and configurable size of PCM buffer added in Linux kernel v5.5 brings better experience to users. The rest of topic comes from comparison to what existent userspace driver, libffado2[11], does. 3. device aggregation Some user pointed that it is not available with drivers in ALSA firewire stack to aggregate several audio data stream into one stream. It is what libffado2 does. Let me describe my opinion about it. At first, let me start with fundamental attribute of audio data frame. In my understanding about ALSA PCM interface, audio data frame is a unit for audio data transmission. The audio data frame includes the specific number of audio data depending on hardware; e.g. 2 samples for usual sound device. The fundamental attribute of audio data frame is to include the set of audio data sampled at the same time. The goal of aggregating audio data stream is to construct an audio data frame from some audio data frames for several devices. It means that one audio data frame consists of audio data sampled at different time. As I describe the phenomena about nominal and effective frequency, each hardware seems to run own unique effective frequency time to time[12], at least over IEEE 1394 bus. Additionally, we have the experience that the hardware is not aware of sequence of packet with nominal frequency for sample synchronization. It might be legitimate that we can not pick up audio data frame which consists of audio data sampled at the same time even if they are transferred at the same isochronous cycle[13]. When achieving the aggregation, we would need to loosen up the fundamental attribute of audio data frame, by accepting the range of sampling time for audio data in the frame, or need to implement one of resampling methods to adjust phase of audio data to the frame. Although the aggregation is itself superficially useful, it seems not to be a requirement to device driver in hardware abstraction layer of general purpose operating system. It may be over engineering. [1] Linux 5.14 https://lore.kernel.org/lkml/CAHk-=wh75ELUu99yPkPNt+R166CK=-M4eoV+F62tW3TVg… [2] The cooperation is done in my public repository in github.com: https://github.com/takaswie/snd-firewire-improve [3] The method is described in the message: [alsa-devel] IEEE 1394 isoc library, libhinoko v0.1.0 release https://lore.kernel.org/alsa-devel/20190415153053.GA32090@workstation/ [4] The resolution of time stamp is 24.576 MHz, which is the same as specification of cycle time in IEEE 1394. The method to compute time stamp of packet for samples in the packet is defined by IEC 61883-6. We can see integrated document for it published by industry group: https://web.archive.org/web/20210216003054/http://1394ta.org/wp-content/upl… [5] I borrow the expression from IEEE 1722. We can see specific term, sampling transmission frequency (STF) in IEC 61883-6 to express similar idea of the media clock. [6] [PATCH 0/3] ALSA: firewire: media clock recovery for syt-aware devices https://lore.kernel.org/alsa-devel/20210601081753.9191-1-o-takashi@sakamocc… [7] [PATCH 0/6] ALSA: firewire: media clock recovery for syt-unaware devices https://lore.kernel.org/alsa-devel/20210531025103.17880-1-o-takashi@sakamoc… [8] [PATCH 0/3] ALSA: firewire-motu: media clock recovery for sph-aware devices https://lore.kernel.org/alsa-devel/20210602013406.26442-1-o-takashi@sakamoc… [9] SNDRV_PCM_HW_PARAMS_NO_PERIOD_WAKEUP. When the PCM substream has a flag of SNDRV_PCM_INFO_NO_PERIOD_WAKEUP, it's available. [10] Precisely except for snd-isight. [11] http://www.ffado.org/ [12] Precisely the hardware looks to run own unique media clock over IEEE 1394 bus. [13] For precise discussion, the knowledge about IEC 61883-6 and vendor specific method for packetization is required. Regards Takashi Sakamoto

3 years, 10 months

B.Oops 1.8.0 - Glitch Sound Effect Sequencer

by Sven Jaehnichen

Hi, a the new version of B.Oops is out now. With really a LOT of new features: * New: Slot shape mode: Controlled by a user-defined shape instead of a pattern * New: Slot keys mode: Controlled by user-defined MIDI events instead of a pattern * New: Pattern randomization * Fx * New Banger * New EQ * Tremolo: Waveform option * Oops: New sample * Default optimization flags `-O3 -ffast-math` for compiling DSP * Improved binary compatibility / portability using static libs * User friendly hiding patterns for inactive slots * New presets * New: Provide binary packages * Bugfixes * Fix pattern Y flip glitches * Correctly X flip merged pads * Fix paste merged pads causing overlaps * Bugfix remove slots may cause segfault * Fix clicks on decay * Bugfix clicked handles if shape changed Thanks to the community for the ideas and suggestions. B.Oops has AFAIK now got more features than any commercial and closed-source effect sequencer! And you can further contribute to this project. Github: https://github.com/sjaehn/BOops Download: https://github.com/sjaehn/BOops/releases/tag/1.8.0 Preview: https://www.youtube.com/watch?v=nHJlSlvxit8 Regards, Sven

3 years, 10 months

PCM channel names API

by Philippe Bekaert

Same context as my previous mail … The RME HDSPe cards have a large number of inputs and outputs for which the standard ALSA channels names and mappings make little sense. Names reflect the hardware interfaces, e.g. Analog.L, Analog.R, AES.1/1 … AES.1/8, ADAT.1 … ADAT.8 etc… and not their function. Right now, the hdspm driver provides a ad-hoc virtual file in /proc/asound/card<n> containing these names. I am looking for a more generic driver <-> user space API for communicating PCM cannel names. Has anyone been facing the same / similar issues? How have you solved it? Are there any ideas, suggestions, preferences concerning this topic? Again looking forward to your feedback. Best regards and thanks in advance, Philippe.

3 years, 10 months

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Linux-audio-dev