Re: [Alsa-user] Problem reading back periods

Paul D. DeRocco wrote: > If I open my sound output device (Intel HDA on Intel mobo), set its > parameters as follows: > > access = SND_PCM_ACCESS_RW_INTERLEAVED > format = SND_PCM_FORMAT_S32_LE > channels = 2 > rate = 48000 > periods = 4 > buffer_size = 1024 > > it returns no errors. If I use

Re: [Alsa-user] Slot assignment not working

Paul D. DeRocco wrote: > options snd slots=snd-soc-hifiberry-dacplus,snd-usb-audio > > midiC0D0 a USB MIDI device > midiC1D0 another USB MIDI device > pcmC2D0p and pcmC2D1p on-board crappy PWM audio > pcmC3D0p Hifiberry DAC > pcmC4D0p

Re: [Alsa-user] Slot assignment not working

Paul D. DeRocco wrote: >> Anyway, that slots options specifies only two slots. >> If you have five devices, it's a better idea to specify five slots. > > What I have is three devices that always exist, one of which is the > Hifiberry DAC that I want to use, and the others are the PWM audio and the

Re: [Alsa-user] Slot assignment not working

Paul D. DeRocco wrote: >> Paul D. DeRocco wrote: >>> options snd slots=snd-soc-hifiberry-dacplus,snd-usb-audio >>> >>> midiC0D0 a USB MIDI device >>> midiC1D0 another USB MIDI device >>> pcmC2D0p and pcmC2D1p on-board crappy PWM audio >>> pcmC3D0p

Re: [Alsa-user] Slot assignment not working

Paul D. DeRocco wrote: > Can this parameter be supplied on the kernel command line as snd.slots=...? Yes; this is possible for all parameters. Regards, Clemens -- Check out the vibrant tech community on one of the

Re: [Alsa-user] Sound BlasterX G1, incorrect sample rate

Aaron Jackson wrote: > I bought a Sound BlasterX G1. The website says it supports 24/96 > but I can't get it to go any higher 16/44.1. Many Creative devices require a vendor-specific command to enable high speed. You'd have to monitor what commands the Windows driver sends, if you know how to do

Re: [Alsa-user] Advice on snd_pcm_writei sizes

Paul D. DeRocco wrote: > If my software naturally produces samples in 64-frame chunks, but the > device is set up to have 256-frame periods, is there any advantage to > accumulating four chunks in my own buffer and writing entire periods, > versus writing each chunk separately and expecting to be

Re: [Alsa-user] A soundcard to record IEC958_SUBFRAME from MiniDisc

Mark Hills wrote: > a soundcard that has toslink input, and supports IEC958_SUBFRAME_* > sample formats In practice, PCI cards with a CMI8738/CMI8768 (not CMI878x) chip. Regards, Clemens -- Check out the vibrant tech

Re: [Alsa-user] Changing device while stream is running

John Z. wrote: >> There is no plugin that can do this dynamically, except "pulse". > > That's a bit unfortunate, as I've put a lot of effort to learn enough > about alsa so that I can be able to remove PA from my system. What is so bad about PA that an ALSA plugin would do better? > I don't mind

Re: [Alsa-user] can play back but can't capture

Kent Tong wrote: > Simple mixer control 'Capture',0 > Front Left: Capture 63 [100%] [30.00dB] [on] > Front Right: Capture 63 [100%] [30.00dB] [on] > Simple mixer control 'Input Source',0 > Items: 'Front Mic' 'Rear Mic' 'Line' > Item0: 'Front Mic' > Simple mixer control 'Front Mic Boost',0

Re: [Alsa-user] can play back but can't capture

Ralf Mardorf wrote: > On Thu, 23 Nov 2017 08:59:44 +0100, Clemens Ladisch via Alsa-user wrote: >>> Please find the aadebug.log below. >> >> I did not find it. > > It isn't below, it's attached. Well, I did look below where attachments would have been shown, but did

Re: [Alsa-user] can play back but can't capture

Kent Tong wrote: > Please find the aadebug.log below. I did not find it. > arecord captures basically silence Probably wrong mixer settings. Regards, Clemens -- Check out the vibrant tech community on one of the

Re: [Alsa-user] can play back but can't capture

Kent Tong wrote: > Below is the output of "arecord -l". I have no idea what that Alt device is. > > card 0: PCH [HDA Intel PCH], device 0: Generic Analog [Generic Analog] hw:0,0 (or better plughw:0,2 for automatic sample format conversion) > card 0: PCH [HDA Intel PCH], device 2: Generic Alt

Re: [Alsa-user] PCM control in alsamixer not working when using asound.conf (dmix)

Tobx wrote: > I want to set the sample rate to 44.1 kHz in dmix in order to prevent > upsampling. Override defaults.pcm.dmix.rate. > Is the Master control a hardware control? That depends on whether your hardware has it. Regards, Clemens

Re: [Alsa-user] [asoundrc] "lossless" format conversion without any rate conversion?

Vincent Yu wrote: > Will any sample rate conversions occur with this .asoundrc config? > > pcm_slave.force_24_bit_no_rate_convert { > pcm "hw:0,0" > format S24_3LE # Or other 24-bit format > rate "unchanged" # Necessary? > } > > pcm.my_new_default { > type plug >

Re: [Alsa-user] aplay underrun length

Samuele Carcagno wrote: > I often get underruns Is some other process blocking the CPU or disk I/O? Try increasing the period size. > a single underrun lasting about 122-ms would be negligible for my > experiment Please note that the stream is stopped, and restarted later. Can you actually

Re: [Alsa-user] Changing device while stream is running

John Z. wrote: > My general problem is: I'd like to be able to switch device that > plays audio, while the audio stream is working - without player > application knowing anything happened. PulseAudio can do this. > My current asoundrc can switch the card through environmental >

Re: [Alsa-user] how to purge old ALSA controls without reboot

frede...@ofb.net wrote: > I'm trying to figure out how to delete unused volume controls which > had appeared in previous versions of my .asoundrc. > > Can this be done without rebooting my computer? Unload the sound driver (rmmod), then remove the entries from asound.state, then re-load the

Re: [Alsa-user] Xonar D2X optical out problem

David Woodfall wrote: > I've have had a Xonar D2X for some time now and normally use coax > spdif output to an SMSL Sanskrit DAC. > [...] > I couldn't get any signal via the optical out at all and there isn't > any light that you normally see shining through the output socket. The coax and

Re: [Alsa-user] Get triggered when the device opens

Frank Thomas via Alsa-user wrote: > The actual sound controller is connected via an I²S interface and > synchronizes on its the word select signal. After the synchronisation, > additional configuration must to occur before you can get valid samples Then why don't you do the configuration in the

Re: [Alsa-user] Get triggered when the device opens

Frank Thomas via Alsa-user wrote: > On Mon, 2018-06-04 at 15:23 +0200, Clemens Ladisch via Alsa-user wrote: >> Frank Thomas via Alsa-user wrote: >>> The actual sound controller is connected via an I²S interface and >>> synchronizes on its the word select signal

Re: [Alsa-user] No sound cards seen with kernel 4.17.2 and 4.17.3

Pete wrote: > Linux phoenix 4.16.16: > 1 [Generic]: HDA-Intel - HD-Audio Generic "Generic" shows that the kernel does not know the HDA controller in your chipset. > Linux phoenix 4.17.3: > --- no soundcards --- There should be an error message in the system log. Regards, Clemens

Re: [Alsa-user] Phonic Helix Board 12 Universal: Recording is silent

Andreas Böhler wrote: > all recordings are completely silent Please show the output of "amixer -c2". Regards, Clemens -- Check out the vibrant tech community on one of the world's most engaging tech sites,

Re: [Alsa-user] No sound cards seen with kernel 4.17.2 and 4.17.3

Peter Chant wrote: > Clemens wrote: >> Pete wrote: >>> Linux phoenix 4.17.3: >>> --- no soundcards --- > >> There should be an error message in the system log. > > Jul 2 19:44:10 phoenix kernel: [ 12.072075] snd_hda_intel :09:00.1: > Handle vga_switcheroo audio client > Jul 2 19:44:10

Re: [Alsa-user] Phonic Helix Board 12 Universal: Recording is silent

Andreas Böhler wrote: > On 02/07/18 23:02, Clemens Ladisch via Alsa-user wrote: >> Andreas Böhler wrote: >>> all recordings are completely silent >> >> Please show the output of "amixer -c2". > > There is no output at all. "alsamixer -c2"

Re: [Alsa-user] Problems opening devices

Robert Bielik wrote: > After this I try snd_pcm_open on the IDs, most of which I get -EBUSY. Did you actually close the device from the previous try? Check in /proc/asound/cardX/pcm0p/sub0/status if the device is opened. Regards, Clemens

Re: [Alsa-user] Strange i/o problem

Robert Bielik wrote: > It works nicely if I either: > 3. Pipe capture -> playback with a larger buffer size, such as 64. It's possible that the hardware does not actually handles size 32 correctly. > The rendering thread is (pseudo code): > > while (true) { > if(capture_active) { >

Re: [Alsa-user] Iec958 plugin usage

(please don't top-post) maruthi srinivas wrote: > By AES, i mean AES standard as in: > https://en.m.wikipedia.org/wiki/AES3 This is a hardware standard; it does not define the format of samples in memory. > This will add 4bit before and 4 bit after pcm sample.it seems iec958 plugin > adds those

Re: [Alsa-user] alsactl sometimes fails to restore mixer settings at boot

Nikos Chantziaras wrote: > alsactl sometimes fails to restore the mixer settings at system boot. Failure > rate is about 50/50. > > systemd-udevd[151]: Process '/usr/sbin/alsactl restore 1' failed with exit > code 99. > systemd-udevd[149]: Process '/usr/sbin/alsactl restore 0' failed with exit

Re: [Alsa-user] dmix with non-hw slave ?

Robert Bielik wrote: > Is it possible on the ALSA side to setup dmix to use the "jack" type > plugin as backend ? No; dmix requires a feature that is supported only by the kernel driver. > Or will the jack plugin do the same job as dmix, i.e. mix together > applications using the device ? Every

Re: [Alsa-user] splitting a 4 channel usb sound card into master and headphone

Samuel Nicholas wrote: > I have a Pioneer DDJ-WeGO, 4 channel USB device that outputs stereo > master and stereo headphone. > I've been scouring the internet trying to find out how I can possibly > split up the ports as it presents as a surround 4.0 device. Try something like this:

Re: [Alsa-user] splitting a 4 channel usb sound card into master and headphone

Samuel Nicholas wrote: > src/conf/cards/USB-Audio.conf > > on lines 22 starts the description of configuring the usb device to be > two stereo devices not a surround40 No, that is for devices that implement two stereo devices in hardware and should be seen as a single 4.0 device by software.

Re: [Alsa-user] alsactl sometimes fails to restore mixer settings at boot

Nikos Chantziaras wrote: > How do I assign numbers? With a line options snd slots=snd-virtuoso,snd-usb-audio in some .conf file in /etc/modprobe.d/. Regards, Clemens -- Check out the vibrant tech community on one

Re: [Alsa-user] MIDI subscription

Paul D. DeRocco wrote: > Under "Subscription", it says "subscription is a connection between two > sequencer ports." The snd_seq_subscribe_port() function takes a > snd_seq_port_subscribe_t which specifies a sender and a receiver. But the > explanation always seems to refer to subscribing "to" a

Re: [Alsa-user] Bower's & Wilkins PX headphone - snd-usb-audio error

a...@philippludwig.net wrote: > usb 3-4: New USB device found, idVendor=19b5, idProduct=0021, bcdDevice=25.20 > usb 3-4: Product: PX USB > usb 3-4: Manufacturer: Bowers & Wilkins > snd-usb-audio: probe of 3-4:1.0 failed with error -22 > usb 3-4: 2:1: cannot get min/max values for control 2 (id 2)

Re: [Alsa-user] MOTU Microbook iic support

jidder--- wrote: > Anyone that could help getting the "Class compliant" Motu Microbook iic to > run under Linux? In theory: plug it in. If something different happens in practice, you have to tell us about it. Regards, Clemens

Re: [Alsa-user] MIDI events of custom USB device not appearing in aseqdump

Kevin Cuzner wrote: > My device presents a USB Midistreaming interface with the following > characteristics: > - A single embedded MIDI IN jack with ID 1 > - A single embedded MIDI OUT jack with ID 2 > - A single element with ID 3 > - The embedded MIDI IN jack is attached to the single input

Re: [Alsa-user] List devices

Robert Bielik wrote: > I cannot get it listed with ALSA API function snd_device_name_hint. This is undocumented; see the "hint" entries in the standard .conf files. Regards, Clemens -- Check out the vibrant tech

Re: [Alsa-user] aplay weirdness

Philip Rhoades wrote: > This produces a crashing static sound: > > espeak --stdout 'words to speak' | aplay > > but this works as expected: > > espeak --stdout 'words to speak' > ./t > aplay ./t Is there a difference in the output of "hexdump -C -n48 ./t" and "espeak --stdout 'words to

Re: [Alsa-user] Help with Audient iD14

Charles Mulder via Alsa-user wrote: > I am unable to select the device using alsamixer. What exactly happens when you try it? Regards, Clemens -- Check out the vibrant tech community on one of the world's most engaging

Re: [Alsa-user] USB Synchronization w/ALSA via g_audio

Rob wrote: > I am capturing data from my microphone, and transmitting it via > pcm_write() to the USB gadget card, and arecording on the USB host. > If I query the buffer, I begin to see a creeping fill on the device > side buffers, such that after a few hours of constant streaming, the > buffer

Re: [Alsa-user] Best course of action with Audient iD14

Charles Mulder wrote: > I would like to ask your help to establish what the best course of action > would be. Ask the PulseAudio developers how to configure it to detect the device correctly: https://www.freedesktop.org/wiki/Software/PulseAudio/Documentation/User/Community/ Regards, Clemens

Re: [Alsa-user] Best course of action with Audient iD14

Charles Mulder wrote: > I am trying to set up a dedicated audio workstation for composing and > would like to keep it as minimal as possible. I was thinking of using > ALSA and Jack without Pulseaudio. > > In your opinion, is that a mistake? Probably not. But the "Analogue Surround 4.0"

Re: [Alsa-user] Iec958 plugin usage

maruthi srinivas wrote: > On Monday, January 22, 2018, Clemens Ladisch > wrote: >> maruthi srinivas wrote: >>> I want to use plugin to convert either way and write to a file. >> >> What for? There is no standard for AES data in a file. >> > Aplay help lists

Re: [Alsa-user] Dmix problem

Robert Bielik wrote: > is there some ALSA plugin that can coalesce buffering ? Meaning that > the plugin can take f.i. larger period_size than what the dmix device > is working with ? What problem would that solve? Regards, Clemens

Re: [Alsa-user] Dmix problem

Robert Bielik wrote: >> Is there any other plugin doing the same thing as dmix... but working ? Yes, dmix with a larger buffer size (i.e., more periods). > I'd need a mixing plugin that does not do sample rate conversion, i.e. each > client connecting to it should be forced to use the mix plugs

Re: [Alsa-user] Iec958 plugin usage

maruthi srinivas wrote: > Iam trying to convert pcm (from wav file) to AES and vice versa. What exactly do you mean with "AES"? > I found a alsa plugin iec958. Will that plugin does do what I want > to achieve ? The iec958 creates an IEC958 bit stream, which is required for S/PDIF on some very

Re: [Alsa-user] USB Audio and selecting optical out

Marc Haber wrote: > On Tue, Apr 24, 2018 at 02:54:44PM +0200, Clemens Ladisch via Alsa-user wrote: >> Marc Haber wrote: >>> On Tue, Apr 24, 2018 at 02:43:01PM +0200, Clemens Ladisch via Alsa-user >>> wrote: >>>> Check if it's actually using 44.1 k

Re: [Alsa-user] USB Audio and selecting optical out

Marc Haber wrote: > Is it possible that the USB device is only able to play back at 48 kHz? Yes, it's possible. Check /proc/asound/cardX/stream0. > On Sun, Apr 08, 2018 at 05:20:13PM +0100, James wrote: >> You might also need to mess with AES0 settings. Google for that. The USB audio driver

Re: [Alsa-user] USB Audio and selecting optical out

Marc Haber wrote: > On Sun, Apr 22, 2018 at 10:45:11PM +0200, Clemens Ladisch via Alsa-user wrote: >> Marc Haber wrote: >>> Is it possible that the USB device is only able to play back at 48 kHz? >> >> Yes, it's possible. Check /proc/asound/cardX/stream0. > >

Re: [Alsa-user] USB Audio and selecting optical out

Marc Haber wrote: > On Tue, Apr 24, 2018 at 02:43:01PM +0200, Clemens Ladisch via Alsa-user wrote: >> Marc Haber wrote: >>> Looks to me like it at least claims to be able to play back at 44.1 kHz. >>> Any other reason why the DAT deck won't sync if speakertest runs with

Re: [Alsa-user] Stereo Output on Asus Xonar DGX

Chad Skeeters wrote: > It plays "Front Left", "Front Right" out of *both* speakers of headphones > connected to the green headphone port on the back of the card. Then it > plays "Rear Left", "Rear Right" out of both speakers of headphones > connected to the black surround port on the back of the

Re: [Alsa-user] ALSA on non-Linux systems

Kevin Smallman wrote: > is that down to the hardware support provided by the kernel, or something > like ALSA or OSS? ALSA consists of three parts: - the userspace library, alsa-lib; - the ALSA framework in the kernel; and - various ALSA drivers, also in the kernel. The library can use other

Re: [Alsa-user] A query on the support status for the headset jack on the Clevo P751DM2-G (and other systems) with the Realtek ALC892+TI Burr Brown OPA2134 operational Amplifier

Dennis Mungai wrote: > One thing I've never been able to get to work, even with hdajackretask ( > https://fossies.org/linux/alsa-tools/hdajackretask/README), is the > headphone jack. We'd have to know how the hardware is wired up. > The oddity here is that this Texas Instruments device (OPA2134

Re: [Alsa-user] Thunderbolt audio driver?

Jacob Juul Klejs Kolding wrote: > Are there currently any Thunderbolt audio devices supported in Linux? >From a software point of view, Thunderbolt is just a PCIe bus. So this depends on what PCI(e) audio chip the device is using. Regards, Clemens

Re: [Alsa-user] CM6206-LX linein mute ?

Luca Toldo wrote: > It works fine when I connect the USB Sound microusb to my MacBook, however > when I try to connect it to the Raspberry, it is mute. Please show the output of "amixer". > https://www.raspberrypi.org/forums/viewtopic.php?f=38=233817=1430727 All three inputs _except_ the line

Re: [Alsa-user] CM6206-LX linein mute ?

Luca Toldo wrote: > Simple mixer control 'PCM Capture Source',0 > Items: 'Mic' 'Line' 'IEC958 In' 'Mixer' Please try all settings. Regards, Clemens ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net

Re: [Alsa-user] Thunderbolt audio driver?

Jacob Juul Klejs Kolding wrote: > All I know is that the main chip is an FPGA. > > Are there any tools for Windows that can snif the communication? For PCI(e), this cannot be done in software. Regards, Clemens ___ Alsa-user mailing list

Re: [Alsa-user] CM6206-LX linein mute ?

The "PCM Capture Source" control is currently set to "Mic"; try the others. Regards, Clemens ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user

Re: [Alsa-user] Driver for ESI Gigaport HD+ for 24/96 capability

Nicolas wrote: > https://www.esi-audio.com/products/gigaporthd+/ > > As I understand it, there is no specific Linux driver for this DAC as there > is one in Windows. The web site claims native CoreAudio supports, so there should not be any secret sauce. Please show the contents of

Re: [Alsa-user] human readable names for audio and midi channels

Stefan Sauer wrote: > when using multi channel audio interfaces, it would be awesome to be > able to label the ports to make it easier to select the right (pair of) > channels in the software. ALSA has no mechanism to associate names to multiple channels belonging to the same PCM device. >

Re: [Alsa-user] Unable to see MIDI keyboard with the aconnect -i command

Guy Tucotte wrote: > The command 'amidi -l' shows the device > BUT, the command 'aconnect -i' doesn't show the device Does it help to re-plug the device? > The snd_rawmidi module as been compiled inside the kernel image. Why only some modules? I don't know if it is related, but please try to

Re: [Alsa-user] USB midi interface named 'USB Device 0x86a'

Stefan Sauer wrote: > 'lsusb' shows the device as: > Bus 006 Device 003: ID 086a:0001 Emagic Soft- und Hardware GmbH Unitor8 > > 'amidi -l' shows a human readable name for all midi devices, but the 'Emagic > Unitor amt 8' (hw:4,0) show up with a generic name: > Dir DeviceName > IO hw:4,0,0

Re: [Alsa-user] alsa not seeing sound cards

Brian J. Murrell wrote: > # aplay -l > aplay: device_list:272: no soundcards found... Are the sound drivers loaded? (see the output of "lsmod") Are the devices accessible? (see "ls -l /dev/snd/") Regards, Clemens ___ Alsa-user mailing list

Re: [Alsa-user] Using the multi plugin with a device with different number of inputs and outputs

Francesco Cameli wrote: > Is there a way to create a multi device using a different input and output > channels number? Only by creating two devices, one for each direction: pcm.multiES8 { type asym playback.pcm { type multi ... } capture.pcm { type multi

Re: [Alsa-user] arecord format-conversion

Alex Vazquez wrote: > pcm.!default { > ... > > arecord -r 16000 -f S16_LE -c 1 -d 3 rec.wav > Recording WAVE 'rec.wav' : Signed 16 bit Little Endian, Rate 16000 Hz, Mono > arecord: set_params:1361: Unable to install hw params: Please check with the -v option whether it actually uses your

Re: [Alsa-user] USB Sound Card & default device?

Hans Malissa wrote: > Can anyone tell me how to configure CMUS to use the Fulla 3 or the internal > sound card, no matter how ALSA sorts the cards? ALSA itself has no built-in mechanism to dynamically select a fallback device. I'd recommend to use PulseAudio:

Re: [Alsa-user] USB audio microphone not working on Kingston Hyperx Amp soundcard

Jesse Hathaway wrote: > On Thu, Dec 12, 2019 at 8:01 AM Clemens Ladisch > wrote: >> Then I cannot explain the "No such file or directory" message. >> What is the output of "strace -e open arecord -D plughw:1 test.wav"? > > openat(AT_FDCWD, "/dev/snd/pcmC0D0c", O_RDWR|O_NONBLOCK|O_CLOEXEC) = -1

Re: [Alsa-user] connecting ALSA from a VM to the host?

Rich Morin wrote: > I'd like to set up ALSA to use the host's audio output system. You have to configure the VM to provide a (virtual) sound hardware device (https://www.virtualbox.org/manual/ch03.html#settings-audio), or to pass through a USB audio device from the host. Regards, Clemens

Re: [Alsa-user] USB audio microphone not working on Kingston Hyperx Amp soundcard

Jesse Hathaway wrote: > I just purchased a Kingston HyperX Amp USB soundcard[1]. The microphone input > does not work correctly in Alsa What exactly does not work? What happens when you run "arecord -D plughw:1 test.wav"? > When I try to choose it with alsamixer alsamixer just shows mixer

Re: [Alsa-user] need hardware suggestion

Dmitri Seletski wrote: > My Xonar DX is dying long painful death(power port is loose). It should be possible to resolder it (for somebody who has experience). > HD 598, cushions break apart and don't seem replacable, somewhat loose cable. Parts that wear out are designed to be replacable:

Re: [Alsa-user] USB audio microphone not working on Kingston Hyperx Amp soundcard

Jesse Hathaway wrote: >>> openat(AT_FDCWD, "/dev/snd/pcmC0D0c", O_RDWR|O_NONBLOCK|O_CLOEXEC) = -1 >>> ENOENT (No such file or directory) Are there ACL restrictions on this file? ("getfacl /dev/snd/pcmC0D0c") Does accessing the device node directly work ("cat /dev/snd/pcmC0D0c"), i.e., is the

Re: [Alsa-user] USB audio microphone not working on Kingston Hyperx Amp soundcard

Jesse Hathaway wrote: > $ arecord -D plughw:1 test.wav > arecord: main:828: audio open error: No such file or directory What is the output of "ls -l /dev/snd/"? Does your user have access rights to all device nodes in there? Regards, Clemens ___

Re: [Alsa-user] USB audio microphone not working on Kingston Hyperx Amp soundcard

Jesse Hathaway wrote: > $ arecord -D plughw:1 test.wav > arecord: main:828: audio open error: No such file or directory What is the output of "arecord -l"? Regards, Clemens ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net

Re: [Alsa-user] USB audio microphone not working on Kingston Hyperx Amp soundcard

Jesse Hathaway wrote: > On Thu, Dec 12, 2019 at 1:35 AM Clemens Ladisch > wrote: >> Does your user have access rights to all device nodes in there? > > yes, I am part of the audio group: > > $ ls -l /dev/snd/ > crw-rw+ 1 root audio 116, 2 Dec 12 05:34 pcmC1D0c Then I cannot explain the "No

Re: [Alsa-user] a mono phone interface

Alfred Ganz wrote: > HW Params of device "usb-phone": What are the capabilities of the actual hardware ("hw:1")? Regards, Clemens ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user

Re: [Alsa-user] Routing hw PCM

Gabriele Zampieri wrote: > My goal is to bridge those streams (|bt-capture -> gsm-playback| and > |gsm-capture -> bt-playback|) in a fullduplex configuration to handle calls. > In fact, the bluetooth will be connected to an handset. > > I was testing different asound.conf configuration, but I

Re: [Alsa-user] a mono phone interface

Alfred Ganz wrote: > I have an old Skype phone that I have used under Centos-6 that I have not > been able to get to work under Centos-7 and Centos-8. I have tested it > under Centos-6 (2.6.32-754.6.3.el6.i686 #1 SMP) with: > aplay -D usb-phone -t wav /usr/share/sounds/alsa/Front_Center.wav >

Re: [Alsa-user] 'Supported bit depths' according to alsa-lib seem to differ from real values

f...@interkatz.de wrote: > according to hda-verb, the supported capturing parameters are 16/20/24 > bits per sample These are the sample formats that are transmitted on the HDA bus. > Alsa-lib, on the other hand, claims that the supported formats (for > `hw:0,0` in this case) are S16LE and S32LE

Re: [Alsa-user] Creative Sound Blaster AE-9 drivers

Ethan Grizzle wrote: > Is it being worked on at all, or is it not a priority at the moment? Neither Creative nor anybody else is writing a driver. Regards, Clemens ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net

Re: [Alsa-user] Low volume from "ALC888-VD Analog" device

D. R. Evans wrote: > Clemens Ladisch via Alsa-user wrote on 2/20/20 12:48 AM: >> Please show the output of "amixer scontents". > > Simple mixer control 'Master',0 > Capabilities: pvolume pswitch pswitch-joined > Playback channels: Front Left - Front Right

Re: [Alsa-user] Low volume from "ALC888-VD Analog" device

D. R. Evans wrote: > Clemens Ladisch via Alsa-user wrote on 2/20/20 1:56 PM: >> Please show the output of "amixer -c0 scontents". > > Output attached (it seemed a bit long to put in the body of the e-mail). Looks OK. Try changing the Front Panel Audio Enable jum

Re: [Alsa-user] Low volume from "ALC888-VD Analog" device

D. R. Evans wrote: > the audio seems to have very low output. Please show the output of "amixer scontents". > this issue was raised 11 years ago "ALC888" does not mean anything; what matters is how the chip is wired up. (If you hardware requires something special to be configured, it's possible

Re: [Alsa-user] Connection of multiple headphones to kodi simultaneously (on rpi3)

Harald Heigl wrote: > if one of my devices (bluetooth headphones turned off) is > failing, the whole configuration (another bluetooth headphone+hdmi) is > failing. Is there an easy way to handle alsa failures of one device > gracefully? Not with ALSA plugins. This should be possible with

Re: [Alsa-user] alsa-only applications (such as jitsi) via jackd

Peter P. wrote: > Can I set pcm device names per application as defaults? If the application does not allow you to set the ALSA device name, you have to replace the device definition that the application used (probably "default"). It would be possible to create a device definition that reads an

Re: [Alsa-user] Capturing audio on second capture input

Christopher Lansing wrote: > I can't record from "Capture 1". How many devices show up in the output of "arecord -l"? Regards, Clemens ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user

Re: [Alsa-user] Capturing audio on second capture input

Christopher Lansing wrote: > card 0: SB [HDA ATI SB], device 2: ALC892 Alt Analog [ALC892 Alt Analog] You have to tell whatever program you're using to use this device. For example, "arecord -D plughw:0,2 -f dat something.wav". Regards, Clemens ___

Re: [Alsa-user] Can I use amidi to send (midi) data to the midi input port on a soft-synth?

Jaime wrote: > Can I use amidi to send this sysex message to linuxsampler? No; amidi works only with raw MIDI ports. Load the snd-virmidi module to get a virtual device that you can connect to linuxsampler. Regards, Clemens ___ Alsa-user mailing

Re: [Alsa-user] RME Exspansion

Maurizio Mileto Principale wrote: > Can you tell me if ALSA plans to convert the AE-9 drivers for Linux in the > near future? There does not exist any public information about Creative hardware. Linux drivers are written by Creative. Or not, when they don't feel like it. Regards, Clemens

Re: [Alsa-user] Rodecaster Pro USB

Nils wrote: > I am trying to get a Rodecaster Pro working, and see it in lsusb, but > can't get it to work with Alsa. > > [ 3490.377201] usb 1-7: parse_audio_format_rates_v2v3(): unable to retrieve > number of sample rates (clock 1) This device appears not to be completely class compliant. Does

Re: [Alsa-user] Android Pixel C in MIDI mode not creating a alsa seq port

Stefan Sauer wrote: > If I use my pixel c tablet, the alsa device is recognized (it shows > up in /proc/asound/ and also creates a midi0 entry) Does it show up in the output of "amidi -l"? Is the ID 5209 or 520a? Your logs show both. Regards, Clemens

Re: [Alsa-user] Android Pixel C in MIDI mode not creating a alsa seq port

Stefan Sauer wrote: > idProduct=5209 is the Pixel C as a midi device, 520a is when it connects > as MTP. Your lsusb output showed 520a. Was it supposed to be in MIDI mode at that time? Please show the output of "lsusb -d 18d1:5209 -v". Regards, Clemens

Re: [Alsa-user] aplay -l doesn't see my loopback sound card using snd_aloop

Jeff Tchang wrote: > WARNING *** > Warning! aplay uses ALSA emulation instead of the native OSS API This message shows that your system is set up to use not the ALSA kernel drivers but the OSS drivers. > I have an embedded system with a custom

Re: [Alsa-user] Xonar D2X: Playback only when other devices generate interrupts

Matthias Leofeine wrote: > Playback is interrupted (only short pieces of sound are played/repeated) > or I've got no sound at all; applications log message: "underrun > occurred". This sounds like the ASM1083 problem, but that affected only PCI cards. The PCI/PCIe bridge on the D2X card is not

Re: [Alsa-user] PCM plugin last

Robert Bielik wrote: >> It might be possible to run dmix on a virtual loopback sound card and route >> that one to the LADSPA plugin > > Can this loopback routing be done entirely within a .asoundrc configuration? No; some program must be running to record from the loopback card and play to the

Re: [Alsa-user] PCM plugin last

Robert Bielik wrote: > I’d like to know if there is any way to get the following sound chain through > ALSA: > > Mediaplayer -> dmix -> LADSPA -> plughw:0,0 The dmix plugin plays shared-memory tricks with the ring buffer and therefore requires to run on top of a hw device. It might be possible

Re: [Alsa-user] documentation for error codes?

D. R. Evans wrote: > Many functions say about the return value: > 0 on success otherwise a negative error code > > Where is the documentation for the various possible "negative error code"s? has a list of error codes defined by

Re: [Alsa-user] New user with an ICUSBAUDIO7D

Alan Corey wrote: > arecord -f S16_LE -d 10 -r 16000 --channels=1 --device="hw:2,0" > /tmp/test-mic.wav > Recording WAVE '/tmp/test-mic.wav' : Signed 16 bit Little Endian, Rate 16000 > Hz, Mono > arecord: set_params:1345: Channels count non available I suspect recording always uses channels.

Re: [Alsa-user] Parallel playback on multiple USB cards

Alexandre Bouillot wrote: > ... Raspberry Pi > > I'm able to address each of the 6 cards independently correctly. > However, when I'm trying to have a second channel - not to mention the third > - working in parallel with the first, I face problems. >From what I've heard, the Pi's USB controller

Re: [Alsa-user] [alsa-user] Error: Unable to install hw params

Vijay Rakesh Munganda wrote: > I had tried to record a sample audio using arecord, “arecord hw:0,6 -d 10 > test.wav” but I got an error as unable to install hw params. Use plughw:0,x instead of hw:0,x to get automatic sample format conversion. Or tell arecord to use a sample rate/format that

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