[SOLVED] Deadbeef: Use alsa:PCM instead of software mixer

Hello,
Does anyone know how to get Deadbeef to use PCM alsa mixer instead of it's internal audio software mixer?
Last edited by diffycat (2014-09-01 16:43:18)

emeres wrote: Have you ever seen such a setup?
No. That's why I'm asking.
emeres wrote:Creating a script that would watch deadbeef volume and change PCM or whatever you specify using amixer. The problem is getting deadbeefs volume. Now-playing switch does not seems to provide that.
I can get volume level with mpris-plugin. The only problem is that it crashes with current ddb version.

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[SOLVED] Software mixing on ALSA+USB sound card

I have cheap chinesse USB soud card. lsusb says its "Tenx Technology, Inc. TP6911 Audio Headset"
After arch installation sound was working about 2-3 reboots. Then i googled and deleted pulseaudio, and sound randomly working or not after reboots.
After a lot of research i got /etc/asound.conf like that:
pcm.!default {
type hw
card AUDIO
ctl.!default {
type hw
card AUDIO
"AUDIO" its name of card. Checked that. And now i had sound in 80% reboots.
After research i founded that i cant play 2 sounds at same time. (And youtube writes "error occured" for 90% videos).
Tested aplay (it never works):
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Playing WAVE '/usr/share/sounds/alsa/Front_Center.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Mono
aplay: set_params:1239: Channels count non available
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So that mean i need software mixing.
Googled, and installed "alsa-oss" and did some random stuff with config
pcm.!default {
type hw
card AUDIO
slave.pcm "dmix"
ctl.!default {
type hw
card AUDIO
slave.pcm "dmix"
and now youtube working. I hear it. But now aplay says:
>aplay /usr/share/skype/sounds/CallBusy.wav
ALSA lib pcm_hw.c:1743:(_snd_pcm_hw_open) Unknown field slave
aplay: main:722: audio open error: Invalid argument
He dont like "slave" thing in config, but it helps. I tested 3 times, added "slave" and rebooted and youtube works. Without "slave" it not working.
And i still cant hear 2 sounds at same time. In google ALSA says like software mixing must work without config, if not then do bugreport.
Im in arch less then 7 days, before i had experience with ubuntu server only, so configuring ALSA is hard for me.
Can somebody help me configure software mixing?
Thank you.
Last edited by pico (2013-07-07 10:36:33)

Okay. I surfed your URLs and ended in http://alsa.opensrc.org/Dmix
Example /etc/asound.conf from 3 and 5 works great! Thank you!
If somebody will google Tenx Technology, Inc. TP6911 Audio Headset
Use /etc/asound.conf (you need create it)
pcm.dsp0 {
type plug
slave.pcm "hw:0"
ctl.mixer0 {
type hw
card 0
or
pcm.dsp0 {
type plug
slave.pcm "dmix"
hint {
show on
description "My dmix dsp0"
ctl.mixer0 {
type hw
card 0
both works for me.
=======
Aaaaand after 2 reboots everything is broken. Both configs not helping. aoss and alsaplayer don't make any sound.
5 reboots later sound again works.
=======
Okay, i know why i need reboot.
If you have same problem - do aplay -l, if your card not first - nothing will work. You need reboot until your card is first. I tryed to do my card always first, but then mixer dont work. And for make it work on every reboot looks like you need read 18 huge books about universe creation.
Last edited by pico (2013-07-09 09:53:25)

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[SOLVED] Asus 900 : ALSA plays sound, but microphone doesn't work

Hi,
This week I installed Arch Linux (kernel 3.1.8-1-ARCH) on my Asus 900 with HDA Intel Realtek ALC662 rev1 sound card. The wiki and google and existing forum posts have been good for most issues, but I'm stumped on ALSA configuration. So here's my first forum post for advice.
I configured sound OK following advice in the wiki and twiddling settings with alsamixer, through both the internal speaker and through the headphones, but I can't get internal or external microphone to work. I'm testing with:
arecord -d 3 junk.wav
then when I play back via:
aplay junk.wav
there is only soft static (with audible soft clicks at the start and end).
(Other files like /usr/share/sounds/alsa/Front_Center.wav play fine.)
alsamixer recognizes and reports my HDA Intel card, Realtek ALC662 rev1.
https://wiki.archlinux.org/index.php/Alsa has several suggestions of options for the snd-hda-intel module.
Different pages suggest different places, which was confusing until I discovered that they can go in any file in /etc/modprobe.d so I'm putting them in /etc/modprobe.d/snd-hda-intel.conf .
I added
options snd-hda-intel enable_msi=1
with no luck.
I have tried various model=XXX parameters (based on many suggestions found googling for Asus 900, alsa, ALC662, etc problems):
options snd-hda-intel model=auto
options snd-hda-intel model=asus
options snd-hda-intel model=asus-laptop
options snd-hda-intel model=laptop
options snd-hda-intel model=ref
options snd-hda-intel model=asus-mode1
options snd-hda-intel model=asus-mode2
options snd-hda-intel model=asus-mode8
and I do
rmomd snd-hda-intel && modprobe snd-hda-intel
after each modification, which makes a little click in the speakers as the module is restarted.
I don't know what the differences between asus-mode1, asus-mode2 etc are supposed to be - I've failed to find clear documentation about them, but AFAIK they seem to be the only valid model choices, unlike stuff like laptop, asus, etc.
I find asus-mod1 etc at https://github.com/torvalds/linux/blob/ … _realtek.c and various other webpages.
I found they vary the items presented by alsamixer but none of the 8 asus-modeN values helped me get the mic working. (The only obvious change was asus-mode8 made my headphone stop working, unlike asus-mod1 through asus-mod7.)
I looked at /var/log/kernel.log when doing the snd-hda-intel module reload for various values of model= and they all look similar to this:
snd_hda_intel 0000:00:1b.0: PCI INT A disabled
snd_hda_intel 0000:00:1b.0: PCI INT A -> GSI 16 (level, low) -> IRQ 16
snd_hda_intel 0000:00:1b.0: irq 43 for MSI/MSI-X
snd_hda_intel 0000:00:1b.0: setting latency timer to 64
hda_code: ALC662 rev1: BIOS auto-probing
input: HDA Digital PCBeep as /devices/pci0000:00/0000:00:1b.0/input/input24
input: HDA Intel Mic as /devices/pci0000:00/0000:00:1b.0/sound/card0/input25
input: HDA Intel Headphone as /devices/pci0000:00/0000:00:1b.0/sound/card0/input26
(Note the emphasized line about BIOS auto-probing only comes up when not using model= or when trying various suggested model values that don't seem to be actually defined for ALC662, e.g. stuff like model=asus, etc.)
Here's a list of alsamixer settings I used for various models. (Note I also tried muting and raising various higher boost levels, and when there's an Internal Mic or F-Mic item I tried various combinations. The following are just what seem the most plausible settings to me.)
model=asus
model=asus-auto
model=auto
alsamixer presents 14 items (boosts are paired) :
Master 100 (dB gain: 0.00)
Headphone 00
Speaker 100 (dB gain: 0.00, 0.00)
PCM 100 (dB gain: 0.00, 0.00)
Mic 56 (dB gain: 0.00, 0.00)
Mic Boost 22 (dB gain: 10.00, 10.00)
Mic Boost 22 (dB gain: 10.00, 10.00)
Beep MM
Capture 14 (dB gain: 0.00, 0.00)
Auto-Mute Mode Enabled
Digital 25 (dB gain: 0.50, 0.50)
Internal Mic 56 (dB gain: 0.00, 0.00)
Internal Mic Boost 22 (dB gain: 10.00, 10.00)
Internal Mic Boost 22 (dB gain: 10.00, 10.00)
model=asus-mode1
alsamixer [All] presents 8 items:
Master 100 (dB gain: 0.00)
PCM 100 (dB gain: 0.00, 0.00)
Mic 56 (dB gain: 0.00, 0.00)
S/PDIF mute
S/PDIF Default PCM 00
Beep MM
Capture 14 (dB gain: 0.00, 0.00)
Digital 25 (dB gain: 0.50, 0.50)
speaker & headphones work, neither mic works
model=asus-mode2
alsamixer [All] presents 11 items:
Master 100 (dB gain: 0.00)
Headphone 00
Speaker 100 (dB gain: 0.00, 0.00)
PCM 100 (dB gain: 0.00, 0.00)
Mic 56 (dB gain: 0.00, 0.00)
S/PDIF mute
S/PDIF Default PCM 00
Beep MM
Capture 14 (dB gain: 0.00, 0.00)
Digital 25 (dB gain: 0.50, 0.50)
F-Mic 56 (dB gain: 0.00, 0.00)
model=asus-mode3
alsamixer [All] presents 9 items:
Master 100 (dB gain: 0.00)
PCM 100 (dB gain: 0.00, 0.00)
Mic 56 (dB gain: 0.00, 0.00)
S/PDIF mute
S/PDIF Default PCM 00
Beep MM
Capture 14 (dB gain: 0.00, 0.00)
Digital 25 (dB gain: 0.50, 0.50)
F-Mic 56 (dB gain: 0.00, 0.00)
model=asus-mode4
alsamixer presents 9 items
Master 100 (dB gain: 0.00, 0.00)
Headphone 100 (dB gain: 0.00, 0.00)
PCM 100 (dB gain: 0.00, 0.00)
Mic 56 (dB gain: 0.00, 0.00)
S/PDIF mute
S/PDIF Default PCM 00
Beep MM
Capture 14 (dB gain: 0.00, 0.00)
Digital 25 (dB gain: 0.50, 0.50)
model=asus-mode5
alsamixer presents same 9 items as asus-mode4
model=asus-mode6
alsamixer [All] presents 9 items same as asus-mode3
model=asus-mode7
alsamixer [All] presents 12 items:
Master 100 (dB gain: 0.00, 0.00)
Headphone1 00
Headphone2 00
Speaker 100 (dB gain: 0.00, 0.00)
PCM 100 (dB gain: 0.00, 0.00)
Mic 56 (dB gain: 0.00, 0.00)
S/PDIF mute
S/PDIF Default PCM 00
Beep MM
Capture 14 (dB gain: 0.00, 0.00)
Digital 25 (dB gain: 0.50, 0.50)
IntMic 56 (dB gain: 0.50, 0.50)
model=asus-mode8
alsamixer [All] presents 11 items:
Master 100 (dB gain: 0.00, 0.00)
Headphone1 00
Headphone2 00
Speaker 100 (dB gain: 0.00, 0.00)
PCM 100 (dB gain: 0.00, 0.00)
Mic 56 (dB gain: 0.00, 0.00)
S/PDIF mute
S/PDIF Default PCM 00
Beep MM
Capture 14 (dB gain: 0.00, 0.00)
Digital 25 (dB gain: 0.50, 0.50)
No headphone output for asus-mode8!?
I'm not interested in doing anything fancy (e.g. mixing multiple sources or whatever), I just want basic mic functionality.
Finally here's some info requested from the archlinux.org/index.php/Alsa:
$ aplay -l
**** List of PLAYBACK Hardware Devices ****
card 0: Intel [HDA Intel], device 0: ALC662 rev1 Analog [ALC662 rev1 Analog]
Subdevices: 1/1
Subdevice #0: subdevice #0
$ lsmod|grep snd
snd_hda_intel 19325 0
snd_hda_codec_realtek 211044 1
snd_hda_codec 69829 2 snd_hda_intel,snd_hda_codec_realtek
snd_hwdep 4942 1 snd_hda_codec
snd_pcm 60207 2 snd_hda_intel,snd_hda_codec
snd_timer 15438 1 snd_pcm
snd 43817 6 snd_hda_intel,snd_hda_codec_realtek,snd_hda_codec,snd_hwdep,snd_pcm,snd_timer
soundcore 5018 1 snd
snd_page_alloc 5869 2 snd_hda_intel,snd_pcm
$ ls -l /dev/snd
total 0
drwxr-xr-x 2 root root 60 Jan 11 20:43 by-path
crw-rw---T 1 root audio 116, 5 Jan 11 20:43 controlC0
crw-rw---T 1 root audio 116, 4 Jan 11 20:43 hwC0D0
crw-rw---T 1 root audio 116, 3 Jan 11 21:01 pcmC0D0c
crw-rw---T 1 root audio 116, 2 Jan 11 21:01 pcmC0D0p
crw-rw---T 1 root audio 116, 1 Jan 11 15:23 seq
crw-rw---T 1 root audio 116, 33 Jan 11 15:23 timer
Has anyone got sound recording working on an Asus 900 with HDA Intel Realtek ALC662 rev1 ...?
Thanks for any help! (And for feedback whether this post gave too much info or not enough, or other suggestions for posting such requests. Sorry if this was too long, but it seemed better to err on the side of too much info...)
Last edited by goulo (2012-01-13 18:30:13)

Hello,
I don't have your soundcard but maybe the following information is helpful, though:
Install the linux-docs package, then you have some information about your soundcard drivers in "/usr/src/linux-*/Documentation/sound/alsa/". Exspecially "HD-Audio.txt" and "HD-Audio-Models.txt" might be important. In the Models file it is stated that just the "asus-mode*" are legitimate model parameters for your soundcard and in the HD-Autio.txt file it is more or less said that trial and error the modules is the way to go to find the "correct module". Further there is a section
Capture Problems
~~~~~~~~~~~~~~~~
The capture problems are often because of missing setups of mixers.
Thus, before submitting a bug report, make sure that you set up the
mixer correctly. For example, both "Capture Volume" and "Capture
Switch" have to be set properly in addition to the right "Capture
Source" or "Input Source" selection. Some devices have "Mic Boost"
volume or switch.
When the PCM device is opened via "default" PCM (without pulse-audio
plugin), you'll likely have "Digital Capture Volume" control as well.
This is provided for the extra gain/attenuation of the signal in
software, especially for the inputs without the hardware volume
control such as digital microphones. Unless really needed, this
should be set to exactly 50%, corresponding to 0dB -- neither extra
gain nor attenuation. When you use "hw" PCM, i.e., a raw access PCM,
this control will have no influence, though.
It's known that some codecs / devices have fairly bad analog circuits,
and the recorded sound contains a certain DC-offset. This is no bug
of the driver.
So check your mixer, here and here are nice tutorials.
Greetings
matse
Last edited by matse (2012-01-12 16:30:09)

[SOLVED] Pulse Audio / ALSA audio sharing.

SOLUTION AT POST # 4
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What can I do to make my audio work as it should?
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00:1b.0 Audio device: Intel Corporation 6 Series/C200 Series Chipset Family High Definition Audio Controller (rev 05)
Screenshot of the only sound card option I have in the XFCE Audio Mixer Plugin:
http://imgur.com/3TJWC
I searched the Wiki, but it is out of date. https://wiki.archlinux.org/index.php/PulseAudio#ALSA
. . . For the applications that do not support PulseAudio and support ALSA it is recommended to install the PulseAudio plugin for ALSA. This package also contains the necessary /etc/asound.conf for configuring ALSA to use PulseAudio.
To prevent applications from using ALSA's OSS emulation and bypassing Pulseaudio (thereby preventing other applications from playing sound), make sure the module snd_pcm_oss is not in the MODULES array in /etc/rc.conf. If it is currently loaded (lsmod|grep oss), disable it by executing: . . .
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Edit: Since there is a lack of response, I am completely changing the entire post in order to be clear and productive.
Last edited by qKUqm3wtY4 (2013-01-05 04:52:15)

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I've just finished setting up ncmpcpp with MPD, and it works great for playing my music and displaying it in the program. I really like the visualizer feature that comes with ncmpcpp, so I looked at the Wiki guide on activating it. On the wiki page, it uses 'mpd.fifo' to send data about the song file (I presume) to enable the visualization. By following the wiki page, I got the visualization working fine, but my audio was no longer working. According to XFCE's Audio Mixer program, I'm using ALSA. I don't really know what this means or why I can't get the visualization working, or if there's another package I could install to bridge the non-working ALSA with the working FIFO (in terms of ncmpcpp's visualizer). How can I enable the visualizations with ALSA?
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Last edited by Zulfe (2015-01-27 08:44:26)

th3voic3 wrote:Could you post your mpd.conf? Remember to put it in code tags.
MPD
music_directory "/home/damon/Music/"
playlist_directory "/home/damon/.mpd/playlists"
db_file "/home/damon/.mpd/mpd.db"
log_file "/home/damon/.mpd/mpd.log"
pid_file "/home/damon/.mpd/mpd.pid"
state_file "/home/damon/.mpd/mpdstate"
audio_output {
type "alsa"
name "My Audio"
path "/tmp/mpd.alsa" #I just assumed that you could replace mpd.fifo with mpd.alsa to use ALSA with the visualizer
format "44100:16:2"
ncmpcpp
mpd_music_dir = "/home/damon/Music/"
mpd_connection_timeout = "5"
mpd_crossfade_time = "5"
playlist_disable_highlight_delay = "0"
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progressbar_color = "cyan"
statusbar_color = "white"
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color2 = "cyan"
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frank604 wrote:I'm confused. First of all, how do you not know what you installed? Did you run an install script? Next is, did you get the visualization working? Post your ncmpcpp config.
I know that I installed ncmpcpp and configured it to work with MPD. MPC is also active and managing my local database. I'm only unsure of what I've installed in terms of the ALSA audio driver. When I went through setting up my system, I downloaded all of the dependencies to get Arch up and running, and I therefore may have installed ALSA in the process when it was instead FIFO that I needed to make this work. The visualization worked fine when I set it up with mpd.fifo, but that cancelled out my audio. I could no longer hear my music. Config above.
Last edited by Zulfe (2015-01-27 18:09:28)

Using Alsa with OSS

Hello everyone. I'm using Alsa with Pulseaudio for sound, but I'm having trouble getting it to work with my computer's speakers. I can, however, get Alsa to work with my usb headset. I considered using OSS instead of Alsa, but I heard that it didn't work with usb microphones. Is there any way to run both Alsa and OSS at the same time, so I can use the headset and the speakers? Thanks.

I think Alsa might not have loaded my drivers
lsmod | grep '^snd' | column -t
Gives me
snd_usb_audio 111545 0
snd_usbmidi_lib 19027 1 snd_usb_audio
snd_rawmidi 18742 1 snd_usbmidi_lib
snd_seq_device 5180 1 snd_rawmidi
snd_hda_codec_hdmi 27857 1
snd_hda_codec_realtek 62247 1
snd_hda_intel 33946 4
snd_hda_codec 101474 3 snd_hda_codec_realtek,snd_hda_codec_hdmi,snd_hda_intel
snd_hwdep 6364 2 snd_usb_audio,snd_hda_codec
snd_pcm 76956 5 snd_usb_audio,snd_hda_codec_hdmi,snd_hda_codec,snd_hda_intel
snd_page_alloc 7298 2 snd_pcm,snd_hda_intel
snd_timer 18687 2 snd_pcm
snd 58893 17 snd_hda_codec_realtek,snd_usb_audio,snd_hwdep,snd_timer,snd_hda_codec_hdmi,snd_pcm,snd_rawmidi,snd_usbmidi_lib,snd_hda_codec,snd_hda_intel,snd_seq_device
By the way, I'm using my Asrock mobo's integrated sound card.

HT201302 The laptop I was using to upload photos from my iphone is no longer good and I want to use my desktop instead but it is displaying an error that the iphone has no photos in it. How do I start using my desktop to upload my photos from my iphone5?

The laptop I was using to upload photos from my iphone is no longer good and I want to use my desktop instead but it is displaying an error that the iphone has no photos in it. How do I start using my desktop to upload my photos from my iphone5, is there any software I need to download?

Photos on the phone in the DCIM folder are those located in the camera roll. Photos that are synced to the phone from the computer cannot be accessed in this manner. They are synced and the only way to manage them is to connect the phone, go to the photos tab and un-select the folders you want to remove or go into the folder photos are located in and remove the photos you no longer want to sync. Then sync the phone and the photos will be removed.

Using the Supra brand, ekey software on BB 10

Good morning all, I have heard that ekey software can be downloaded and used on the Q10, who has done this and how? Thank you and have a good morning.

Please don't double post. I answered your question already:
http://supportforums.blackberry.com/t5/BlackBerry-Q10/how-to-use-the-Supra-Brand-ekey-software-on-my...
1. Please thank those who help you by clicking the "Like" button at the bottom of the post that helped you.
2. If your issue has been solved, please resolve it by marking the post "Solution?" which solved it for you!

[SOLVED] Deadbeef: Use alsa:PCM instead of software mixer

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