Advanced Linux Sound Architecture Activity
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Activity for Advanced Linux Sound Architecture
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James Lehman modified a comment on discussion alsa-user
Hello everyone. I have been using modified multichannel sound devices to produce laser control signals for many years. Easily the most common device to be found are these inexpensive USB surround sound boxes that contain some variation of the CMedia USB sound chip. I have four different devices and I am using an Orange Pi Zero 3 4gb running dietpi with the latest 6.6 kernel. But that doesn't seem to matter. I get the same results any other way I try it. The strange thing is that one of my devices...
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Hello everyone. I have been using modified multichannel sound devices to produce laser control signals for many years. Easily the most common device to be found are these inexpensive USB surround sound boxes that contain some variation of the CMedia USB sound chip. I have four different devices and I am using an Orange Pi Zero 3 4gb running dietpi with the latest 6.6 kernel. But that doesn't seem to matter. I get the same results any other way I try it. The strange thing is that one of my devices...
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Philip Brown modified a comment on discussion alsa-user
I am looking for some help with a popular device in the ham radio world that uses the Burr-Brown USB Audio CODEC from TI. It is brand named "Signalink USB" by Tigertronics. Here is what I get when it is plugged in: pi@raspberrypi:~ $ aplay -l * List of PLAYBACK Hardware Devices * card 1: CODEC [USB Audio CODEC], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 pi@raspberrypi:~ $ arecord -l * List of CAPTURE Hardware Devices * card 1: CODEC [USB Audio CODEC], device 0: USB...
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I am looking for some help with a popular device in the ham radio world that uses the Burr-Brown USB Audio CODEC from TI. It is brand named "Signalink USP" by Tigertronics. Here is what I get when it is plugged in: pi@raspberrypi:~ $ aplay -l * List of PLAYBACK Hardware Devices * card 1: CODEC [USB Audio CODEC], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 pi@raspberrypi:~ $ arecord -l * List of CAPTURE Hardware Devices * card 1: CODEC [USB Audio CODEC], device 0: USB...
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Damien Bourg posted a comment on discussion Help
(I also post here because I am not sure if it was seen in the alsa-user forum) Hello, I bought a soundcard from Raspiaudio, the MIC ULTRA+, https://raspiaudio.com/produit/ultra , which seems to be a great addition to my Raspberry Pi zero (I wish to install Darkice+Icecast to broadcast sounds from my garden). I installed the soundcard successfully following the right method. I can also successfully record with the built in microphones and playback the recordings. However when I plug in an external...
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Hello, I bought a soundcard from Raspiaudio, the MIC ULTRA+, https://raspiaudio.com/produit/ultra , which seems to be a great addition to my Raspberry Pi zero (I wish to install Darkice+Icecast to broadcast sounds from my garden). I installed the soundcard successfully following the right method. I can also successfully record with the built in microphones and playback the recordings. However when I plug in an external microphone (working mic) to the card I can't record and have the following outputs:...
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Hello, I bought a sound card from Raspiaudio, the MIC ULTRA+, https://raspiaudio.com/produit/ultra , which seems to be a great addition to my Raspberry Pi zero (I wish to install Darkice+Icecast to broadcast sounds from my garden). I installed the sound card successfully following the right method. I can also successfully record with the built in microphones and playback the recordings. However when I plug in an external microphone (working mic) to the card I can't record and have the following outputs:...
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Hello, I bought a sound card from Raspiaudio, the MIC ULTRA+, https://raspiaudio.com/produit/ultra , which seems to be a great addition to my Raspberry Pi zero (I wish to install Darkice+Icecast to broadcast sounds from my garden). I installed the sound card successfully following the right method. I can also successfully record the built in microphones and playback the recordings. However when I plug in an external microphone (working mic) to the card I can't record and have the following outputs:...
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Vinnie Moscaritolo posted a comment on discussion Help
Hello; I am developing a car-radio app in C++ (open source) for raspberry pi; https://github.com/vinthewrench/carradio I am having some success with the ALSA api and the Signstek USB DAC, but I would like some advice on how to solve a problem related to using 4 speakers. My code only know about left and right channels at the moment and does snd_pcm_writei calls to the “default” device. I believe I can just add another USB DAC to give me 4 channels (front and back) but I am unclear on how to write...
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David Diamante modified a comment on discussion alsa-user
I am having the same problem using mido library in python on a raspberry pi 4b. Sorry I cant be of help but would love to hear updates on this! pi@diamanted:~/therepi $ python Python 3.9.2 (default, Feb 28 2021, 17:03:44) [GCC 10.2.1 20210110] on linux Type "help", "copyright", "credits" or "license" for more information. >>> import mido >>> mido.get_output_names() ALSA lib seq_hw.c:466:(snd_seq_hw_open) open /dev/snd/seq failed: No such device Traceback (most recent call last): File "src/_rtmidi.pyx",...
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I am having the same problem using mido library in python on a raspberry pi 4b. Sorry I cant be of help but would love to hear updates on this! pi@diamanted:~/therepi $ python Python 3.9.2 (default, Feb 28 2021, 17:03:44) [GCC 10.2.1 20210110] on linux Type "help", "copyright", "credits" or "license" for more information. >>> import mido >>> mido.get_output_names() ALSA lib seq_hw.c:466:(snd_seq_hw_open) open /dev/snd/seq failed: No such device Traceback (most recent call last): File "src/_rtmidi.pyx",...
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I am having the same problem using mido library in python on a raspberry pi 4b. Sorry I cant be of help but would love to hear updates on this!
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Kishan duhatra created ticket #266
aplay: pcm_write:2053: write error: Input/output error
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Frank Portegies posted a comment on ticket #265
Exactly the same problem here with Ubuntu 20.04. I hope that there will be a linux driver soon for the sound for J4115 chip. I want to follow the news about this, if anyone knows ,more let us know! Reinstall of Windows with Kuu unattended installation (on USB stick) worked and gave me sound, But I screwed up the windows registration key that is stored and controlled in the bios. Hardware vendors and Microsoft are working better together to prevent use of illegal windows versions.
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Paljas posted a comment on discussion Help
So, I found an answer to my second question. By querying verb 0xA- (converter format): hda-verb /dev/snd/hwC0D0 0x6 0xa00 0x0 (have to pad 8 trailing bits with 0 due to hda-verb constant command format i.e. verb size), I get confirmation that the card is really sending out the expected sample rate and bit format. While I don't know for sure still if the driver is really telling me the truth, I do see the value change etc. and the HDA specification tells me: "The Converter Format control determines...
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Hi, and apologies if this is not the right place for this question. When I play a 16b .wav over SPDIF (an ALC889 chip), all is OK. When I try to play 24b, I get (wav generated myself with Audacity): aplay -D hw:0,1 --dump-hw-params sine_220_10s_2c_44k1_s24b.wav -v ... aplay: set_params:1233: Sample format non available Available formats: - S16_LE - S32_LE - IEC958_SUBFRAME_LE But the ALC889 datasheet says it supports 24b samples. When I use the plughw instead to allow conversion, it ouputs sound,...
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retnev posted a comment on discussion alsa-user
Ok, Yet again crickets. this group is dead. If anyone lands at this alsa dead door and have the same problem with these interfaces, search for my thread in Ardour UG where I managed to solve the problem the hard way and step over a lot of obvious alsa bugs which never get sorted out. Cheers.
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Alex posted a comment on discussion Help
Hi, all!! For functionality reasons, I need to configure different rates for playback and recording. For capture I need force a rate of 16000 but for playblack I want to keep the default. If I use individually aplay or arecord it works correctly but simultaneously I get errors and it does not work. Would it be correct to use different frequencies for playback and capture for the same card? /etc/asound.conf pcm.!default { type asym playback.pcm { type hw card 0 device 0 } capture.pcm { type plug slave...
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I also want to know why this happens suddenly. Also happens to my Motu gear
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Alsa works great with my UMC1820 interface. Absolute BLISS !! I recently bought the ADA8200 ADAT extension to get 8 more inputs. It is all connected correctly as I get all inputs and outputs and pass sound on all inputs in Windows as a test. However on Linux, alsa only sees 8 inputs on the UMC1820 and only two out of eight inputs on the ADA8200, namely inputs 7&8 works. Inputs 1-6 does not work. Since the UMC1820 makes you select between SPDIF (technically channels 9&10) and the ADAT extension (8...
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I also want to know why this happens suddenly.
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Marco Bernardi created ticket #265
No sound with Intel Smart Sound Tecchnology (ISST) - Intel J4115
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John T. Sylvanis modified a comment on discussion Help
Hi. It is absolutely maddening how hard it is to create an account for ALSA Wiki to download ALSA's components. I cannot pass beyond the TOKEN. Can anyone help, PLEASE? Thx, John.
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Hi. It is absolutely maddening how hard is to create an account for ALSA Wiki to download ALSA's components. I cannot pass beyond the TOKEN. Can anyone help, PLEASE?
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JimR posted a comment on discussion Help
Fedora 33 KDE spin. I recently bought a USB web cam with built in mic. Works fine on Windows 10 on same machine (dual-boot). When I select the mic using alsamixer, I am able to move the volume up to mid-level. But when I use Zoom for a meeting, and start talking, within a few seconds the mic level drops to 0 and meeting participants can't hear me. I tried using Audacity, and it seems to keep the audio level fine for doing a sound recording. Where to look, what to try?
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Charles Hudson posted a comment on discussion Help
Apologies for asking a simple questions but I am lost after new install of Fedora KDE. The machine is a dual-boot ASUS / AMD and has on-board audio, as well as a Xonar DG card and a Hauppage TV tuner. ALSA has arbitrarily selected front-panel headphones for sound output. I would like to change this to use the Xonar card's stereo line out, as is configured in the Win10 applications. This output goes to an amp and speakers. ALSA Mixer doesn't even show this as an option. What am I missing? Thanks for...
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Severin Landwein created ticket #55
MixPre-10 II is not recognized correctly
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Pieter Cardoen posted a comment on discussion alsa-user
Dear We are developing an audio system for an industrial environment with multiple output channels (>10). We want to be able to play audio on all combinations of these audio channels (i.e play on channel 0,1 and 5 or play on 3,4,6 and 8). We can play audio on multiple PCMs by using a pcm of type multi. However, it should be possible to address any combination which means that we have > 1000 combinations. Is it possible to play audio on multiple PCMs in alsa without configuring the asound.conf and...
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andrey volodin created ticket #264
need help with implementing virtual microphone with introduced delay
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Fritz Menzer posted a comment on discussion alsa-user
Hello, We've got a MOTU UltraLite AVB and are trying to get it to work properly with Ubuntu 20 as a class-compliant USB 2.0 device. However, when sound is played to the device, the channel associations switch in regular intervals. For example when you play a stereo signal, the sound may appear on the device in the "From Computer" channels 1/2 as expected, but sometimes it's 17/18 or 9/10 (the jump is always by a multiple of 8 channels). Also, sometimes the sound appears in two pairs of channels,...
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Francisco Fernandez posted a comment on discussion alsa-user
I have a usb sound card "Lexicon Omega". I have used this sound card for years with Ubuntu Studio without any issues. It used to work out of the box. I was able to record 4 channels at the same time. Until I updated to Ubuntu Studio 20.04 On Ubuntu Studio 18.04.4 with kernel 4.15.0-20, Lexicon Omega works great. No problems at all. On Ubuntu Studio 18.04.4 with kernel 4.15.0-101 (and later), Lexicon Omega can only capture 2 channels. Or at least I can only see 2 capture channels in qjackctl. Any...
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This is not really a straight forward user question and thereby I post it in the developer forum. My Problem: I use Alsa/Jack/Pulseaudio to run quite an extensive audio interface to DAW with involved Jack routings of both Audio and Midi. The system runs like greased lightning and I can do things on Linux that I could never do on windows period. The problem is that there exist applicatio9ns that is not ALSA/JACK Midi friendly. One such Application is Bitwig for Linux 1) Bitwig absolutely requires...
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Geordon Marchak created ticket #54
Fender Amp USB feed
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Rezzonics modified a comment on discussion alsa-user
This is the ltrace: root@hardkernel-odroidc2:~# ltrace arecord -D plughw:0,3 -c 1 -f S24_LE -d 1 -t wav -r 48000 /tmp/test.wav __libc_start_main([ "arecord", "-D", "plughw:0,3", "-c"... ] <unfinished ...=""> setlocale() = <void> textdomain() = <void> snd_pcm_info_sizeof() = <void> snd_pcm_info_sizeof() = <void> memset(0xffffdcfac6f0, 0, 288) = 0xffffdcfac6f0 snd_output_stdio_attach() = <void> strstr() = <void> fileno() = <void> isatty() = <void> getopt_long() = <void> getopt_long() = <void> __errno_location()...
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gdb trace: root@hardkernel-odroidc2:~# gdb --args arecord -D plughw:0,3 -c 1 -f S24_LE -d 1 -t wav -r 48000 /tmp/test.wav GNU gdb (GDB) 8.3.1 Copyright (C) 2019 Free Software Foundation, Inc. License GPLv3+: GNU GPL version 3 or later http://gnu.org/licenses/gpl.html This is free software: you are free to change and redistribute it. There is NO WARRANTY, to the extent permitted by law. Type "show copying" and "show warranty" for details. This GDB was configured as "aarch64-poky-linux". Type "show...
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This is the ltrace: root@hardkernel-odroidc2:~# ltrace arecord -D plughw:0,3 -c 2 -f S16_LE -d 1 -t wav -r 48000 /tmp/test.wav __libc_start_main([ "arecord", "-D", "plughw:0,3", "-c"... ] <unfinished ...=""> setlocale() = <void> textdomain() = <void> snd_pcm_info_sizeof() = <void> snd_pcm_info_sizeof() = <void> memset(0xfffffaf0ee90, 0, 288) = 0xfffffaf0ee90 snd_output_stdio_attach() = <void> strstr() = <void> fileno() = <void> isatty() = <void> getopt_long() = <void> getopt_long() = <void> __errno_location()...
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I have solved some issues, audio-routing was missing on device tree, son cpu and codec playback and capture stremas where not properly connected, but arecord does not work yet. there is no default nor sysdefault card when I run arecord -L: root@hardkernel-odroidc2:~# arecord -L null Discard all samples (playback) or generate zero samples (capture) usbstream:CARD=S905 GXBB ODROID-C2 S905 USB Stream Output it should look like aplay -L: root@hardkernel-odroidc2:~# aplay -L null Discard all samples (playback)...
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Chris Keller posted a comment on discussion Help
With some help, I was able to identify a solution. It's PulseAudio that's actually doing hotplug support on my system, so I was able to disable hotplug behavior there. In /etc/pulse/default.pa, find and comment out lines related to module-switch-on-connect: #.ifexists module-switch-on-connect.so #load-module module-switch-on-connect #.endif Now the default audio sink won't change automatically, it must always be done in pavucontrol or the system volume widget.
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root@hardkernel-odroidc2:~# strace arecord -D hw:0,3 -c 1 -f S16_LE -d 10 -t wav -r 48000 /tmp/test.wav execve("/usr/bin/arecord", ["arecord", "-D", "hw:0,3", "-c", "1", "-f", "S16_LE", "-d", "10", "-t", "wav", "-r", "48000", "/tmp/test.wav"], 0xffffd58f4668 / 15 vars /) = 0 brk(NULL) = 0xaaaaefbfd000 faccessat(AT_FDCWD, "/etc/ld.so.preload", R_OK) = -1 ENOENT (No such file or directory) openat(AT_FDCWD, "/etc/ld.so.cache", O_RDONLY|O_CLOEXEC) = 3 fstat(3, {st_mode=S_IFREG|0644, st_size=11901, ...})...
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I have modified play driver and developed capture driver for meson S905 GX device (Odroid-C2) based on mainline Linux v5.6. I have also modified CS4245 codec driver. "play" works fine: root@hardkernel-odroidc2:~# aplay -l * List of PLAYBACK Hardware Devices card 0: S905 [GXBB ODROID-C2 S905], device 0: fe.dai-link-0 () [] Subdevices: 1/1 Subdevice #0: subdevice #0 root@hardkernel-odroidc2:~# aplay -D plughw:0,0 -c 1 -f dat /usr/share/sounds/alsa/Front_Right.wav Playing WAVE '/usr/share/sounds/alsa/Front_Right.wav'...
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Yanone posted a comment on discussion Help
Hi everyone, I'm working on a Raspberry Pi home cinema setup that has an equalizer and a compressor already running successfully. Now I want to add some sort of VU meter (this is all on a headless Pi) to assess the compression levels, both reduction as well as makeup. I'm a beginner to ALSA. I thought it should be possible, since devices in ALSA can be freely chained together, to add two separate loopback devices into the chain, one before the compressor, and one after. Then record sound levels between...
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Peter Kövesdi posted a comment on discussion Help
Hey, I'm running XBian on a rasperry pi. I'd like to have squeezelite running in parallel to Kodi, both using the same soundcard output at the same time. Now, further more I want squeezelite to play directly, and Kodi to play with swapped stereo channels. For this purpose I wrote the following /etc/asound.conf: pcm.dmixer { type dmix ipc_key 1023 ipc_perm 0666 slave { pcm "hw:1,0" rate 44100 } bindings { 0 0 1 1 } } pcm.swapped { type plug slave { pcm "dmixer" } ttable.0.1 1 ttable.1.0 1 } Kodi is...
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Hi folks, I'm using an amatuer radio audio interface called a SignaLink USB; it presents itself as an external sound card and can send and receive audio through the radio. It's working well, but every time I plug it in, the OS picks it up and assumes that I'd like to use the external sound card as the default and starts playing all system sounds through the radio. Is there a way to configure ALSA to not use the SignaLink as the default, ever? Ideally this would only apply to the SignaLink, but I'd...
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Latebo Querty modified a comment on discussion Help
Hi I would like to get some help, because I have difficulties to make a good configuration for my alsa installation. I'm having a Raspberry pi 3 and a HifiBerry AMP2 soundcard. I could set up the basic asound.conf to have everything working, but I have difficulties to set up a more complex example. I would like to ask for your help to try to fix it. I have set up the latest version of Raspbian Lite version. I have installed shairport-sync. Technical infos about the setup ======================================================...
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Hi I would like to get some help, because I have difficulties to make a good configuration for my alsa installation. I'm having a Raspberry pi 3 and a HifiBerry AMP2 soundcard. I could set up the basic asound.conf to have everything working, but I have difficulties to set up a more complex example. I would like to ask for your help to try to fix it. I have set up the latest version of Raspbian Lite version. I have installed shairport-sync. Technical infos about the setup uname -a Linux Konyha 4.19.97-v7+...
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Luis Alvarez modified a comment on discussion Open Discussion
I had troubles using headsets in Ubuntu (18.04 and 19.10). After too many tries, the solution was to modify the file: /etc/modprobe.d/alsa-base.conf appending the line: options snd-hda-intel model=,dell-headset-multi Even though I solved this issue, I ask still why isn't it configured by default. I can give any information that is needed to correct this. Credits to the video: https://youtu.be/00fhAW7qYQk A comment says that the 44th line is also missing: options snd-hda-intel position fix=1 In my...
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I had troubles using headsets in Ubuntu (18.04 and 19.10). After too many tries, the solution was to modify the file: /etc/modprobe.d/alsa-base.conf appending the line: options snd-hda-intel model=,dell-headset-multi Even though I solved this issue, I ask still why isn't it configured by default. I can give any information that is needed to correct this. Thanks
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Richard Hall posted a comment on discussion Help
I've been developing an application in python on the Raspberry Pi and trying to get 100% clean audio from a microphone. Various attempts using the Pyaudio and sounddevice modules give a variety of clicks and pops due to dropped samples. I have also tried spawning both arecord and ffmpeg as sub-processes and then using a named pipe to read the output into my app. These last two were more successful until I increased the number of other threads in the app when once again I started getting dropped samples....
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pK posted a comment on discussion Help
I finally resoled this issue. I had to enable "Build IDT/Sigmatel HD-audio codec support" under: │ -> Device Drivers │ │ -> Sound card support (SOUND [=y]) │ │ -> Advanced Linux Sound Architecture (SND [=m]) │ │ -> HD-Audio I had previously been avoiding it because it never mentions snd_hda_codec_idt but rather snd_hda_codec_sigmatel. Would be great if the description in "make menuconfig's" help section also mentioned SND_HDA_CODEC_IDT for the symbol instead of just SND_HDA_CODEC_SIGMATEL.
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Hi, I'm trying to resurrect an old Hewlett Packard Mini 1000 laptop that I have but wasn't using. I'd like to turn it into a tiny internet radio media box, but I'm struggling to get sound working on it. This is a Gentoo install with alsa-utils 1.1.8 Any assistance with this would be greatly appreciated. I'm about to lose my mind as I've been working to resolve the sound issue for at least two weeks now. Sound works if I boot from a Debian live DVD, but it's also using an kernel 4.19.67-2+deb10u1....
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Cassiano Reinert Novais posted a comment on discussion Help
I'm new in this forum. I have an issue with audio in a DELL Inspiron 14 Laptop that follow me for more than 1 year. I have an extensive documentation about the problem, as I've being gathered along this 1 year. I would to know if I can start a new topic with my specific problem, or isn't Help Forum the right place, and there is another place to post it. Best regards, Cassiano
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Kamal Kishor Pandey posted a comment on discussion Open Discussion
I went through [http://www.pogo.org.uk/~mark/trx/streaming-desktop-audio.html] about streaming desktop audio over network, but I am still not successful in receiving any audio to the receiver side. My “tx” binary is running in a debian 9 desktop with x86 architecture. The contents of my “/etc/asound.conf are as follows: defaults.pcm.!card 1 defaults.ctl.!card 1 defaults.pcm.!device defaults.ctl.!device and the contents of ~/.asoundrc are as follows: pcm.!default { type plug slave.pcm { type dmix...
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Chris Hodgetts posted a comment on discussion alsa-user
Hello, I am wanting to run 4 CM108 USB audio dongles on a single server. I am not sure if this an ALSA or a USB issue but am hoping the group could point me in the right direction. When the machine reboots, the card allocation to the plughw number changes at random depending. I have tried to use udev to allocate a specific name to each card based on the USB port it was attached to, however parts of the final software were not keen on using some of the functions because of the card name change. Basically,...
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ankita jain posted a comment on discussion Help
Hi, I am trying to access USB MIC, UA MIC and USB HP UA HP , together at the same point of time and to transmit different waveforms in diffrent output devices. Presently I can access either of the one Input devices or either of the one output devices through appliction whereas I want both the input and output devices to be accessed at a time. My application code is running successfully when device is chosen as default. If instead of default I am putting "plughw 0:0", I am getting the error as ALSA...
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I guess alsa isnt actively developed anymore ?
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Marcel Brouillet posted a comment on discussion alsa-user
Hello, I have enabled my Linux Ubuntu to be a bluetooth A2DP sink. It works : I press play on my phone and sound comes out of my laptop. I explain in the footer how I did. I only see the devices in PulseAudio, not in Alsa. I understand the Alsa integration has been isolated in a BlueAlsa project. GOAL : I want the devices to show up in Alsa, as my user application (mixxx.org) lists devices found in Alsa. WORKING : I have compiled BlueAlsa from git, and followed the README.md. I disabled pulseaudio...
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sheila posted a comment on discussion alsa-user
Hi! I’m a beginner in all forms of audio processing, so please just let me know if there is some information I’m missing that I should provide (or if this is the wrong place to be, forum wise). I’m running a speech test that uses full duplex through a delta44 connected to my computer (a dell OptiPlex 7040, if that helps). However, whenever I try to use snd_pcm_readi/writei I end up with underrun Alsa buffers where only 1 frame gets read on the first iteration, and then on the next iteration, it generates...
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Gautham P posted a comment on discussion alsa-devel
Calling snd_pcm_hw_params * APIs from the currnet program will provide proper values related to Hardware. However, when these API's are called tru shared library, it is not providing proper valeues. Furthermore, if i include these APIs as static library and call from current program, it still works fine, but when you include as shared lib issue araises! Have any faced such problem? Thanks in Advance, Gautham.P
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TurGu posted a comment on discussion alsa-user
Hello all, I'm trying to use both a Roland HP piano and a Roland A88 midi controller using a USB connection. The os is Debian with kernel 4.9 In a nutshell: The midi keyboard is connected to the computer (odroid-n2 sbc); The command 'lsusb' shows the device The command 'amidi -l' shows the device I can see the device special file /dev/midi1 I can see the midi keyboard output using 'cat </dev/midi1' BUT, the command 'aconnect -i' doesn't show the device The aconnect is important to work such that...
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I am trying to get Alsa to route spdif from an 1818VSl. The alsamixer says Clock sour for spdif. I can use windows and use spdif, so it is not connection problems or hardware problems, it is definately the interaction between 1818vsl and alsa that is the problem. I tried both internal clock and spdif clock in the 1818vsl, but nothing seems to remove the clock-sour message in alsa to get spdif to work in linux. Any ideas what i can still try.
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Ronnie Bailey posted a comment on discussion Open Discussion
Motherboard is an Asus Crosshair VI Hero with onboard audio. Operating System is openSUSE Tumbleweed. snd-hda-intel is what loads as a driver. I am able to set VLC to use specifically use alsa, and further specify the 7.1 version of the card, but VLC is the only package that has sound. Nothing else in my system can utilize the device. The alsa-info script sees the device as card 1, generic device. alsa-info.sh output can be found at: http://alsa-project.org/db/?f=2b8c4b4c6d323e46c4bf15f4ca74fef0639c91cd...
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Aditya Paluri posted a comment on discussion alsa-user
Hey jesse, have you figured out what the issue is .. Seems like I am facing with similar issue at my end too.
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Ali Momeni posted a comment on discussion alsa-user
I'm working on an Nvidia Jetson Nano, on an application that requires midi. It appears that the required ALSA module (snd_seq) is not available on the kernal. As a result, when I run my application, I see: ALSA lib seq_hw.c:466:(snd_seq_hw_open) open /dev/snd/seq failed: No such file or directory MidiInAlsa::initialize: error creating ALSA sequencer client object. terminate called after throwing an instance of 'RtMidiError' what(): MidiInAlsa::initialize: error creating ALSA sequencer client object....
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Geoff Kaniuk posted a comment on discussion alsa-user
Hello Alsa List, Is there a way for alsa to report the maximum allowed voltage on a PC microphone input? I am running Audacity on a dual-booted (Debian Jesie) Dell Inspiron 1150. Audacity has a linear signal scale with range -1 to 1. I am interested to know what voltage range this corresponds to if I record with full volume. I hope that in one of the alsa commands there is a way to get this information. Any help on where to look would be much appreciated.
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I guess this group is dead.
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I have no problem with alsa routing to and from all 8 inputs and 18 outputs on my 1818vsl. However spdif which is usually on inputs 9&10 doesnt work. The spdif setup works as with the windows software all works, but in linux with alsa there is no signal reported on channels 9&10 which alsa does provide for the 1818vsl. Are there special consideration regarding alsa with spdif I am not aware of, Does it need clock timings if so, hwere is the clock timing that alsa get from the 1818vsl. I see no such...
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I have no problem with alsa routing to and from all 8 inputs and 18 outputs on my 1818vsl. However spdif which is usually on inputs 9&10 doesnt work. The spdif setup works as with the windows software all works, but in linux with alsa there is no signal reported on challes 9&10 which alsa does provide for the 1818vsl. Are there special consideration regarding alsa with spdif I am not aware of, Does it need clock timings if so, hwere is the clock timing that alsa get from the 1818vsl. I see no such...
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pc athome created ticket #263
Unable to get AudigyRX to work
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Daniel Bender posted a comment on discussion Open Discussion
Hi, guys. I see you all talking about using the Tascam US 428 in Linux, but I haven't figured out how to do it yet. I'd greatly appreciate any help with getting it working. I use Mint Cinnamon 18.3, and right now have the USB light on in my Tascam unit, but still can't get any sound from it. Please give me some simple ideas about what to do to make this device work, for I'm a novice in Linux configurations. Thanks, Danny
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K AJAY KUMAR modified a comment on discussion alsa-devel
Hi all, can anyone tell me the way of jack detection process of tlv320aic32x family in linux kernel?
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Hi all, can anyone tell me the way of jack detection process of tlv320aic32x family in linux kernel?
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tom favell posted a comment on discussion Open Discussion
Hello, Im running alsaloop test to test the audio on our harware. Im specifically running this command - alsaloop -C hw:0,0 -P hw:1,0 -S 1 -l 480 -c 6 -v When I measure the delay through my hardware I get 11mSec - which is fine, and the audio quality is good as well. However after several hours of running this loop test the delay has increased to 90mSec. Can someone advice what is going wrong here please.
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Park Jin Soo posted a comment on discussion Open Discussion
Oh!! Thanks!!
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Jaroslav Kysela posted a comment on discussion Open Discussion
Fixed in http://git.alsa-project.org/?p=alsa-lib.git;a=commit;h=fdc4c17e1c9f4f7591c7fb4f2bc2e0c399d2156e .
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Is there any progress? Or do I need to ask about this to another mail address? I think no-one care these pages.
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Benedikt posted a comment on discussion Help
Okay, I've figured out a fix. I set Audacity to record at 48 kSps instead of the usual 44.1 kSps. No crackle then Unfortunately, since I need my recordings at 44.1 kSps, I have to resample them but at least it works now.
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Hello everyone, I need help with my Samson Meteor Mic USB microphone, which I have connected to my Raspberry Pi (running Raspbian). I'd like to use it for recording, but I keep getting an occasional crackle in the recording. Just brief clicks but they do make the recording useless. I normally use Audacity, but I've tried the arecord utility, too - same result. It sounds like the data stream cutting out for an instant every few seconds. The microphone works fine on my windows laptop, and also on my...
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c243473 posted a comment on discussion Help
Hi, I am developping an application with asoundlib. I have a usb audio device that I would like to send usb control transfers, but I did not find a way to do so with asoundlib. More specifically I need to send the GET_MEM (https://www.usb.org/sites/default/files/audio10.pdf, p67 - section 5.2.1.2) request which is an audio class specific usb request. Is there a way to do that with the ALSA Library API? Or do I need to modify the snd-usb-audio driver? Thank you!
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I’ve got very basic problems for this function, about snd_ctl_elem_set_bytes. I’m trying to write 512 bytes data using snd_ctl_elem_set_bytes functions. But it’s failed. / \brief Set values to given data as an element of bytes type. \param obj Data of an element. \param data Pointer for byte array. \param size The number of bytes included in the memory block. / void snd_ctl_elem_set_bytes(snd_ctl_elem_value_t obj, void *data, size_t size) { assert(obj); assert(size < ARRAY_SIZE(obj->value.bytes.data));...
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diya posted a comment on discussion Open Discussion
Hello, I am trying to record the songs through alsa arecord and playing it back through aplay command. The recorded song (from ADC stored in processor Memory) when played through aplay seems to run fast But if I play some pre-stored sample through aplay seems to be fine. Audio codec of our design supports 32K sample rate. It would be great if anyone can direct or suggest me on this issue.
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Eric Egberts posted a comment on discussion Open Discussion
Hello, I am trying to route the audio from a webradio station to a program that generates an FM signal on the Raspberry. I have a service that opens a webradio stream using mplayer and outputs it to the default alsa output device: Service1: mplayer -ao alsa -prefer-ipv4 http://77.72.149.48:8108 Then another service grabs the audio from the loopback and sends it to stdin. Then a program that generates a FM signal grabs the audio from stdout. Service2: arecord -D hw:1,1 -f S16_LE -c 2 -r 44100 - |...
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Mika Filpus created ticket #724
HDSPmixer segfaults with RME 9632 + AI4S-192 AIO
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TITUSRATHINARAJ STALIN posted a comment on discussion Open Discussion
Hello, I have used the below command to create audio record loopback device to receive the audio data from BLE. sudo modprobe snd-aloop And now "arecord -l" output looks like below. pi@raspberrypi:~ $ arecord -l * List of CAPTURE Hardware Devices * card 1: Loopback [Loopback], device 0: Loopback PCM [Loopback PCM] Subdevices: 8/8 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 Subdevice #4: subdevice #4 Subdevice #5: subdevice #5 Subdevice...
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RME 9632 has capabilty of 16 inputs but RMEmixer shows only 12. When trying to access those missing four channels by using RME's AI4S analog input boards the mixer app gives segmentation fault. The mixer is fine again when the ribbon cable is disconnected. However, despite of the RMEmixer, all 16 channels are seen and received by Ardour finely.
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Eric Zaluzec modified a comment on discussion alsa-devel
I've modified a Linux kernel sound driver to use a new virtual address instead of a physical address when receiving data. When the driver is called on, it reports that the data is 0. I'm looking for any input or feedback on what I may be missing in my process and implementation. If my descriptions below are too vague to give proper feedback, I am happy to expand and provide more details. I have a physical register at address0xc0056014. The hardware register is readable and is correctly initialized...
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I've modified a Linux kernel sound driver to use a new virtual address instead of a physical address when receiving data. When the driver is called on, it reports that the data is 0. I'm looking for any input or feedback on what I may be missing in my process and implementation. If my descriptions below are too vague to give proper feedback, I am happy to expand and provide more details. I have a physical register at address0xc0056014. The hardware register is readable and is correctly initialized...
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I've modified a Linux kernel sound driver to use a new virtual address instead of a physical address when receiving data. When the driver is called on, it reports that the data is 0. I'm looking for any input or feedback on what I may be missing in my process and implementation. If my descriptions below are too vague to give proper feedback, I am happy to expand and provide more details. I have a physical register at address 0xc0056014. The hardware register is readable and is correctly initialized...
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Sebastien posted a comment on discussion Help
Hi, Finaly seems that all the channel are muted on, specially the cpature one : [root@localhost ~]# amixer -c 0 cset iface=MIXER,name='CAPTURE feedback Playback Volume' unmute numid=9,iface=MIXER,name='CAPTURE feedback Playback Volume' ; type=INTEGER,access=rw---R--,values=2,min=0,max=255,step=0 : values=0,0 | dBscale-min=42949621.21dB,step=0.25dB,mute=1 All the channel have the "mute=1" tag but I'm still unable to unmute all the channels as you can notice the command used. Anybody have an idea how...
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Hi, I've found the "same issue" in a report, the driver and teh card are the same as mine, only the linux flavor differ, mine is CentOS. http://www.alsa-project.org/db/?f=31f5faf68bc25689441d585204868cb57016df31 Is there a known bug fort this card/driver combination, the results are that teh capture could not be toggle on (un mute), neither using alsamixer interface or amixer commands. Thanks
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Hi, Anybody have an idea or a suggestion ? Thanks ! Seb
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Jamey Kirby posted a comment on discussion alsa-user
If I capture using arecord one hour using s24_le sampling, the resulting file is around 5.5 gig. If I capture with ecasound using the exact same sampling parameters, the file is a little over 4 gig. The differenceis over 1 gig in file size. The file from arecord is almost impossible to convert to mp3 file. With the resulting file from ecasound, mp3 are produced without issues. Any idea what is going on? I am using the same hardware on both captures; Behringer UMC 404HD connected to USB under Ubuntu...
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Hi! Havins a strange issue here on a old centos 5/Alsa config for a Sound Blaster Audigy SE card using the CA0106 driver. I'm unable to toggle or to unmute the capture in the mixer, he never get activated so I'm streaming only one song, the sound of silence.. ;-) I think the issue is here, the Capabilites seems to be wrong, I think I sould see cvolume as the capture volume and note the pvolume as the playback one : Simple mixer control 'CAPTURE feedback',0 Capabilities: pvolume Playback channels:...
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I am runnig arecord in continuious capture mode with arecord creating a new file every hour. We use this as an archiving system for our non-profit radio station. We record everything on the air and save it for future reference, or web streaming. I've noticed an issue: arecord keeps files open forever until program exits. This is not good. What happens in a week or two when I have run out of file handles? I've downloade the source and will look into how to close the file each time a new file is created....
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Alan Bromborsky posted a comment on discussion Help
I have install a xonar essence stx ii 7.1 on my ubuntu 16.04 htpc. I was using the analog line in and the spdif (rca) output. Initally both worked. I did a standard ubuntu upgrade and neither worked. Reverting the kernel did not help. My system is dual boot with win 10. The sound card works on win 10. I tested the card (both when it worked and when it didn't) by playing music with kodi. I explicitly selected the stx spdif output device with the kodi system menu. I am looking for any suggestions to...
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Dejan Jovasevic posted a comment on discussion Open Discussion
The issue was not in the way I edited the ALSA config itself, but the way user space application was getting device hints from ALSA. I was looping through sound cards and then for each card I would call snd_device_name_hint (int card, const char *iface, void ***hints). and get back array of hints for all devices on that card. However, when I changed the logic and instead of looping through cards I called the fucntion with -1 snd_device_name_hint(-1,....) , I got back array with hints for all cards...
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Harald Schuster posted a comment on discussion alsa-user
Hi I am using an embedded system with two different audio codecs and one of the codecs is the TI TLV320 which is not working as hoped. I use it in combination with the PjProject to make a VoIP call. When the call starts i often get an underrun and the codec is than prepared (snd_pcm_prepare) and newly started (snd_pcm_start). This works fine when I use the hw plugin from alsa. Due to the fact that I need left and right channel i use the dshare plugin. With this plugin my device often stops playback....
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Hi there, I was wondering if it is possible to edit /etc/asound.conf in such a way as to create alias names for specific devices, and then when a user space application enumerates audio devices have these aliases show up? My goal is to be able to create an alias for 2 hdmi playback devices (outputdev1, outputdev2 for instance), and then use this alias when searching for alsa devices in the application. ie. outputdev1 => plughw:CARD=X,DEV=Y outputdev2 => plughw:CARD=X,DEV=Z I have been able to create...
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Hi there, I was wondering if it is possible to edit /etc/asound.conf in such a way as to create alias names for specific devices, and then when a user space application enumerates audio devices have these aliases show up? My goal is to be able to create an alias for 2 hdmi playback devices (outputdev1, outputdev2 for instance), and then use this alias when searching for alsa devices in the application. ie. outputdev1 => plughw:CARD=X,DEV=Y outputdev2 => plughw:CARD=X,DEV=Z I have been able to create...
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Hi there, I was wondering if it is possible to edit /etc/asound.conf in such a way as to create alias names for specific devices, and then when a user space application enumerates audio devices have these aliases show up? My goal is to be able to create an alias for 2 hdmi playback devices (outputdev1, outputdev2 for instance), and then use this alias when searching for alsa devices in the application. ie. outputdev1 => plughw:CARD=X,DEV=Y outputdev2 => plughw:CARD=X,DEV=Z I have been able to create...
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Joel created ticket #262
alsa on OSX possible?
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