[SOLVED] ALSA no sound interrnal speaker with HDA Intel PCH

Hi, i have no sound from the internal speaker of my laptop but the headphones work.
configuration : http://www.alsa-project.org/db/?f=68a08 … 6d1a1242ae
i have tried to install the driver from realtek and to add the option "model=auto", "model=generic", "model=3stack" to the snd-hda-intel" module without success.
the command "aplay -vv -D hw:0,0 /usr/share/sounds/alsa/test.wav" plays a sound only through the headphones.
have you any idea to help me ?
thanks
Last edited by walkyrie (2012-06-02 13:22:49)

This seems pretty similar to my issue... No sound from my from headphone output at all... I do get sound from the rear output BUT it stops when I plug in the front one!
Ran the monitor option on the python script and see the following...
PLUG HEADPHONES IN:
======================================
Diff for codec 0/2 (0x11060441):
+++
@@ -220,17 +220,16 @@
Control: name="Line-Out Jack", index=0, device=0
Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1
Amp-Out vals: [0x00 0x00]
Pincap 0x0001001c: OUT HP EAPD Detect
EAPD 0x2: EAPD
Pin Default 0x01014010: [Jack] Line Out at Ext Rear
Conn = 1/8, Color = Green
DefAssociation = 0x1, Sequence = 0x0
- Pin-ctls: 0x40: OUT
Unsolicited: tag=0x02, enabled=1
Power: setting=D0, actual=D0
Connection: 1
0x18
Node 0x25 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out
Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1
Amp-Out vals: [0x80 0x80]
Pincap 0x0000001c: OUT HP Detect
======================================
PULL HEADPHONES OUT:
======================================
Diff for codec 0/2 (0x11060441):
+++
@@ -220,16 +220,17 @@
Control: name="Line-Out Jack", index=0, device=0
Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1
Amp-Out vals: [0x00 0x00]
Pincap 0x0001001c: OUT HP EAPD Detect
EAPD 0x2: EAPD
Pin Default 0x01014010: [Jack] Line Out at Ext Rear
Conn = 1/8, Color = Green
DefAssociation = 0x1, Sequence = 0x0
+ Pin-ctls: 0x40: OUT
Unsolicited: tag=0x02, enabled=1
Power: setting=D0, actual=D0
Connection: 1
0x18
Node 0x25 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out
Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1
Amp-Out vals: [0x80 0x80]
Pincap 0x0000001c: OUT HP Detect
======================================
Not a lot of difference, but enough to see that the system sees the event...
Now I need to know what to do to make sure the sound starts on the front headphone output when the rear one is disconnected, or if they are parallel, not have the sound stop ;-)

Similar Messages

[Solved] Alsa, no sound after hibernation

Hi,
Since few weeks, after resume from hibernate, I've not got sound.
I've to reboot the computer to have sound.
My kernel is 3.9.4.
# aplay -l
**** Liste des Périphériques Matériels PLAYBACK ****
carte 0: PCH [HDA Intel PCH], périphérique 0: VT1802 Analog [VT1802 Analog]
Sous-périphériques: 1/1
Sous-périphérique #0: subdevice #0
carte 0: PCH [HDA Intel PCH], périphérique 2: VT1802 Alt Analog [VT1802 Alt Analog]
Sous-périphériques: 1/1
Sous-périphérique #0: subdevice #0
carte 0: PCH [HDA Intel PCH], périphérique 3: HDMI 0 [HDMI 0]
Sous-périphériques: 1/1
Sous-périphérique #0: subdevice #0
How can I fix it ?
Thanks in advance.
Last edited by NeanderMarcl (2013-06-15 11:53:01)

chu887 wrote:
NeanderMarcl wrote:
Hi,
Since few weeks, after resume from hibernate, I've not got sound.
I've to reboot the computer to have sound.
My kernel is 3.9.4.
# aplay -l
**** Liste des Périphériques Matériels PLAYBACK ****
carte 0: PCH [HDA Intel PCH], périphérique 0: VT1802 Analog [VT1802 Analog]
Sous-périphériques: 1/1
Sous-périphérique #0: subdevice #0
carte 0: PCH [HDA Intel PCH], périphérique 2: VT1802 Alt Analog [VT1802 Alt Analog]
Sous-périphériques: 1/1
Sous-périphérique #0: subdevice #0
carte 0: PCH [HDA Intel PCH], périphérique 3: HDMI 0 [HDMI 0]
Sous-périphériques: 1/1
Sous-périphérique #0: subdevice #0
How can I fix it ?
Thanks in advance.
Does your headphone jack work? I have the same problem with the speaker , no sound from it at all, but the headphone works as normal.
My device is :
$ aplay -l
**** List of PLAYBACK Hardware Devices ****
card 0: Intel [HDA Intel], device 0: STAC9228 Analog [STAC9228 Analog]
Subdevices: 0/1
Subdevice #0: subdevice #0
card 0: Intel [HDA Intel], device 1: STAC9228 Digital [STAC9228 Digital]
Subdevices: 1/1
Subdevice #0: subdevice #0
Headphones don't work

No sound with HDA Intel HDMI

I've got no sound with the Intel HDMI (work only with headphone)
Here are some hardware informations :
I've run the alsa-info.sh and my informations are there --> http://www.alsa-project.org/db/?f=82540 … 847ccaaccc
[root@archeee ~]# 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
card 0: Intel [HDA Intel], device 1: ALC662 rev1 Digital [ALC662 rev1 Digital]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 1: HDMI [HDA ATI HDMI], device 3: ATI HDMI [ATI HDMI]
Subdevices: 1/1
Subdevice #0: subdevice #0
[root@archeee ~]# lsmod | grep snd
snd_seq_dummy 1099 0
snd_hda_codec_atihdmi 2051 1
snd_seq_oss 25304 0
snd_seq_midi_event 4452 1 snd_seq_oss
snd_hda_codec_realtek 179443 1
snd_seq 42628 5 snd_seq_dummy,snd_seq_oss,snd_seq_midi_event
snd_seq_device 4313 3 snd_seq_dummy,snd_seq_oss,snd_seq
snd_pcm_oss 33693 0
snd_mixer_oss 14810 1 snd_pcm_oss
snd_hda_intel 18921 0
snd_hda_codec 56728 3 snd_hda_codec_atihdmi,snd_hda_codec_realtek,snd_hda_intel
snd_hwdep 5102 1 snd_hda_codec
snd_pcm 57479 3 snd_pcm_oss,snd_hda_intel,snd_hda_codec
snd_timer 16117 2 snd_seq,snd_pcm
snd 43847 11 snd_seq_oss,snd_hda_codec_realtek,snd_seq,snd_seq_device,snd_pcm_oss,snd_mixer_oss,snd_hda_intel,snd_hda_codec,snd_hwdep,snd_pcm,snd_timer
soundcore 5007 1 snd
snd_page_alloc 5809 2 snd_hda_intel,snd_pcm
DAEMONS Section in rc.conf :
DAEMONS=(syslog-ng network netfs crond hal alsa)
The PC is an eeebox EB1006
http://www.productwiki.com/asus-eeebox-pc-eb1006/
Things I checked :
done a alsamixer and seleted HDA ATI HDMI, enabled sound with the m key
┌───────────────────────────── AlsaMixer v1.0.22 ────────
│ Card: HDA ATI HDMI F1: Help
│ Chip: ATI R6xx HDMI F2: System information
│ View: F3:[Playback] F4: Capture F5: All F6: Select sound card
│ Item: S/PDIF Esc: Exit
│
│
│
│
│
│
│
│ ┌──┐
│ │OO│
│ └──┘
│ < S/PDIF >
│
I've readed a lot of thread with the ATI HDMI sound probleme, and my config seems good, but still there is no sound.
Last edited by drev (2010-02-26 17:17:10)

I seem to be having the same problem playing HDMI audio. Do you happen to remember how you got this resolved?
For any interested parties, my alsa mixer screen looks something like this as well:
──────────────────────────── AlsaMixer v1.0.24.2 ─────────────────────────────
Card: HD-Audio Generic F1: Help
Chip: ATI R6xx HDMI F2: System information
View: F3:[Playback] F4: Capture F5: All F6: Select sound card
Item: S/PDIF Esc: Exit
┌──┐
│OO│
└──┘
< S/PDIF >
Audio works fine through the Realtek chip (HDA ATI SB card). Just not through HDMI.

[Solved] ALSA "This sound device does not have any controls" XDJ-R1

Hey guys,
I have a XDJ-R1 which includes a 24bit 44.1khz USB soundcard that can be used to record audio from the mixer.
When I plug it in it's detected fine. However using alsamixer (or amixer) I don't see any audio controls for volume.
If I start pulseaudio it will detect the device just fine. However, at 100% volume the audio seems rather low. I'm seeing about -24db while my mixer shows the output at around +2db. I suspect the ALSA volume is rather low. But have no way of changing it as I can't seem to see any controls!
Here are some command outputs:
alpaly -l
**** List of PLAYBACK Hardware Devices ****
card 0: Intel [HDA Intel], device 0: ALC269 Analog [ALC269 Analog]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 1: XDJR1 [PIONEER XDJ-R1], device 0: USB Audio [USB Audio]
Subdevices: 1/1
Subdevice #0: subdevice #0
aplay -L
null
Discard all samples (playback) or generate zero samples (capture)
default:CARD=Intel
HDA Intel, ALC269 Analog
Default Audio Device
sysdefault:CARD=Intel
HDA Intel, ALC269 Analog
Default Audio Device
front:CARD=Intel,DEV=0
HDA Intel, ALC269 Analog
Front speakers
surround40:CARD=Intel,DEV=0
HDA Intel, ALC269 Analog
4.0 Surround output to Front and Rear speakers
surround41:CARD=Intel,DEV=0
HDA Intel, ALC269 Analog
4.1 Surround output to Front, Rear and Subwoofer speakers
surround50:CARD=Intel,DEV=0
HDA Intel, ALC269 Analog
5.0 Surround output to Front, Center and Rear speakers
surround51:CARD=Intel,DEV=0
HDA Intel, ALC269 Analog
5.1 Surround output to Front, Center, Rear and Subwoofer speakers
surround71:CARD=Intel,DEV=0
HDA Intel, ALC269 Analog
7.1 Surround output to Front, Center, Side, Rear and Woofer speakers
default:CARD=XDJR1
PIONEER XDJ-R1, USB Audio
Default Audio Device
sysdefault:CARD=XDJR1
PIONEER XDJ-R1, USB Audio
Default Audio Device
front:CARD=XDJR1,DEV=0
PIONEER XDJ-R1, USB Audio
Front speakers
surround40:CARD=XDJR1,DEV=0
PIONEER XDJ-R1, USB Audio
4.0 Surround output to Front and Rear speakers
surround41:CARD=XDJR1,DEV=0
PIONEER XDJ-R1, USB Audio
4.1 Surround output to Front, Rear and Subwoofer speakers
surround50:CARD=XDJR1,DEV=0
PIONEER XDJ-R1, USB Audio
5.0 Surround output to Front, Center and Rear speakers
surround51:CARD=XDJR1,DEV=0
PIONEER XDJ-R1, USB Audio
5.1 Surround output to Front, Center, Rear and Subwoofer speakers
surround71:CARD=XDJR1,DEV=0
PIONEER XDJ-R1, USB Audio
7.1 Surround output to Front, Center, Side, Rear and Woofer speakers
iec958:CARD=XDJR1,DEV=0
PIONEER XDJ-R1, USB Audio
IEC958 (S/PDIF) Digital Audio Output
amixer -c 1
(output is empty)
cat /proc/asound/devices
2: [ 0- 0]: digital audio playback
3: [ 0- 0]: digital audio capture
4: [ 0- 0]: hardware dependent
5: [ 0] : control
6: [ 1- 0]: raw midi
7: [ 1- 0]: digital audio playback
8: [ 1- 0]: digital audio capture
9: [ 1] : control
33: : timer
(with pulseaudio on) pacmd list-sources
3 source(s) available.
index: 2
name: <alsa_input.usb-Pioneer_PIONEER_XDJ-R1_PIONEER_XDJ-R1-00-XDJR1.analog-stereo>
driver: <module-alsa-card.c>
flags: HARDWARE DECIBEL_VOLUME LATENCY
state: SUSPENDED
suspend cause: IDLE
priority: 9049
volume: front-left: 65536 / 100% / 0.00 dB, front-right: 65536 / 100% / 0.00 dB
balance 0.00
base volume: 65536 / 100% / 0.00 dB
volume steps: 65537
muted: no
current latency: 0.00 ms
max rewind: 0 KiB
sample spec: s24le 2ch 44100Hz
channel map: front-left,front-right
Stereo
used by: 0
linked by: 1
fixed latency: 99.95 ms
card: 1 <alsa_card.usb-Pioneer_PIONEER_XDJ-R1_PIONEER_XDJ-R1-00-XDJR1>
module: 5
properties:
alsa.resolution_bits = "24"
device.api = "alsa"
device.class = "sound"
alsa.class = "generic"
alsa.subclass = "generic-mix"
alsa.name = "USB Audio"
alsa.id = "USB Audio"
alsa.subdevice = "0"
alsa.subdevice_name = "subdevice #0"
alsa.device = "0"
alsa.card = "1"
alsa.card_name = "PIONEER XDJ-R1"
alsa.long_card_name = "Pioneer PIONEER XDJ-R1 at usb-0000:00:1d.1-2, full speed"
alsa.driver_name = "snd_usb_audio"
device.bus_path = "pci-0000:00:1d.1-usb-0:2:1.0"
sysfs.path = "/devices/pci0000:00/0000:00:1d.1/usb2/2-2/2-2:1.0/sound/card1"
udev.id = "usb-Pioneer_PIONEER_XDJ-R1_PIONEER_XDJ-R1-00-XDJR1"
device.bus = "usb"
device.vendor.id = "08e4"
device.vendor.name = "Pioneer Corp."
device.product.id = "0170"
device.product.name = "PIONEER XDJ-R1"
device.serial = "Pioneer_PIONEER_XDJ-R1_PIONEER_XDJ-R1"
device.string = "front:1"
device.buffering.buffer_size = "26448"
device.buffering.fragment_size = "6612"
device.access_mode = "mmap"
device.profile.name = "analog-stereo"
device.profile.description = "Analog Stereo"
device.description = "PIONEER XDJ-R1 Analog Stereo"
alsa.mixer_name = "USB Mixer"
alsa.components = "USB08e4:0170"
module-udev-detect.discovered = "1"
device.icon_name = "audio-card-usb"
ports:
analog-input: Analog Input (priority 10000, latency offset 0 usec, available: unknown)
properties:
active port: <analog-input>
What might I be missing? Why can PA figure the device out but ALSA can't?
Last edited by EvanPurkhiser (2014-04-16 09:27:13)

I'm interested in recording from the mixer over USB.
The device also supports playback into the mixer, but I assume this is designed for use in conjunction with using the device as a MIDI-Controller, which I don't have any intentions of doing.
What I've been doing to check the levels of the device is to play a track through the device (making sure that the VU meter for the channel is peaking at around 0db), record using `ffmpeg -f pulse -i 1 recording.wav` and then running `ffmpeg -i recording.wav -af "volumedetect" -f null /dev/null`
Which looks something like this:
[Parsed_volumedetect_0 @ 0x1f0eb80] n_samples: 10265206
[Parsed_volumedetect_0 @ 0x1f0eb80] mean_volume: -26.1 dB
[Parsed_volumedetect_0 @ 0x1f0eb80] max_volume: -12.0 dB
[Parsed_volumedetect_0 @ 0x1f0eb80] histogram_12db: 21
[Parsed_volumedetect_0 @ 0x1f0eb80] histogram_13db: 238
[Parsed_volumedetect_0 @ 0x1f0eb80] histogram_14db: 1619
[Parsed_volumedetect_0 @ 0x1f0eb80] histogram_15db: 7703
[Parsed_volumedetect_0 @ 0x1f0eb80] histogram_16db: 29370
I actually _can_ record directly from alsa, but I have to tell it the format, rate, and channels of the device.
[netbook/scratch/]> arecord -D hw:1 test.wav
Recording WAVE 'test.wav' : Unsigned 8 bit, Rate 8000 Hz, Mono
arecord: set_params:1233: Sample format non available
Available formats:
- S24_3LE
[netbook/scratch/]> arecord -D hw:1 -f S24_3LE test.wav
Recording WAVE 'test.wav' : Signed 24 bit Little Endian in 3bytes, Rate 8000 Hz, Mono
arecord: set_params:1239: Channels count non available
[netbook/scratch/]> arecord -D hw:1 -f S24_3LE -c 2 test.wav
Recording WAVE 'test.wav' : Signed 24 bit Little Endian in 3bytes, Rate 8000 Hz, Stereo
Warning: rate is not accurate (requested = 8000Hz, got = 44100Hz)
please, try the plug plugin
[netbook/scratch/]> arecord -D hw:1 -f S24_3LE -c 2 -r 44100 test.wav
Recording WAVE 'test.wav' : Signed 24 bit Little Endian in 3bytes, Rate 44100 Hz, Stereo
The volume is still just as low as recording with PA though

[Solved] ALSA - Play Audio As Is With No Conversion

Hello all,
I have an issue with ALSA I hope you can help me sort out.
For the sake of this post, I'll refer to everything as if played by mplayer2, since the output is much easier this way.
Basically, I want to play everything "as-is", in regards to samplerate and format.
That is, playing music at 16bit/44.1Khz as such, playing movie audio at 16bit/48Khz as such, playing HD audio at 24bit/96Khz as such and so on.
Now, my first attempt was trying to access the HW directly using mplayer's --ao=alsa:device=hw=0.0 and not limiting the format.
Both samples played at correct sample rate but were outputted at 16 bit, since the hardware supposedly do not support floating point, and the output reverted to default.
Selected audio codec: FLAC (Free Lossless Audio Codec) [libavcodec]
AUDIO: 44100 Hz, 2 ch, s16le, 0.0 kbit/0.00% (ratio: 0->176400)
[AO_ALSA] Format floatle is not supported by hardware, trying default.
AO: [alsa] 44100Hz 2ch s16le (2 bytes per sample)
Selected audio codec: Uncompressed PCM [pcm]
AUDIO: 96000 Hz, 2 ch, s24be, 4608.0 kbit/100.00% (ratio: 576000->576000)
[AO_ALSA] Format floatle is not supported by hardware, trying default.
AO: [alsa] 96000Hz 2ch s16le (2 bytes per sample)
Note the Format floatle is not supported by hardware part.
This seemed a bit weird as using alsa without accessing the hardware directly (going through vmix), the output is indeed floating point.
That is, playing with --ao=alsa I get the following (obviously everything is sampled to the default 48Khz):
Selected audio codec: FLAC (Free Lossless Audio Codec) [libavcodec]
AUDIO: 44100 Hz, 2 ch, s16le, 0.0 kbit/0.00% (ratio: 0->176400)
AO: [alsa] 48000Hz 2ch floatle (4 bytes per sample)
Selected audio codec: Uncompressed PCM [pcm]
AUDIO: 96000 Hz, 2 ch, s24be, 4608.0 kbit/100.00% (ratio: 576000->576000)
AO: [alsa] 48000Hz 2ch floatle (4 bytes per sample)
Note that the output format is floatle.
So, I tried specifying --ao=alsa:device=hw=0.0 --format=floatle, only for it to fail as in the first case and reverting to default 16 bit and correct sample rate.
Specifying the required format directly works for the flac sample (s16le, same as the default) but fails for the HD sample (s24be), as expected - I can remember from my Windows days not so long ago I had to output 24 bit audio padded as 32 bit for it to work.
This lead me to my next attempt: --ao=alsa:device=hw=0.0 --format=s32le, which as far as I understand is 32 bit, but not floating point, which isn't exactly what I need.
Anyway, both files played fine with this mode for the few seconds I tried (assuming the remaining bits were padded with zeros), but real-life files (such as movie soundtrack) produced noticeable audio distortions (cracks and pops of all kinds).
So, I'm back to where I began.
Seems like I need to specify the hardware (device=..._) to get correct sample rate, but then I can't get correct format, or let it go via dmix and then the sample rate is wrong.
Now, having giving it a bit more thought, I realized dmix is outputting anything successfully as floating point, so I guess it pads whatever bits needed with zeros, which is just fine.
Assuming the above is correct, is it possible to write something in my .asoundrc file so that input samplerate is preserved?
I'm not very optimistic as it kind of against the whole idea of software mixing, but who knows..
If there's any way of convincing ALSA that my card is perfectly fine when being fed with floating point directly (device=...), that's an option too.
Bottom line, is there a way to make everything sent to card as-is?
That was long... sorry
Thanks, Adam.
Last edited by adam777 (2012-10-03 07:45:17)

adam777 wrote:
Hello all,
I have an issue with ALSA I hope you can help me sort out.
For the sake of this post, I'll refer to everything as if played by mplayer2, since the output is much easier this way.
Basically, I want to play everything "as-is", in regards to samplerate and format.
That is, playing music at 16bit/44.1Khz as such, playing movie audio at 16bit/48Khz as such, playing HD audio at 24bit/96Khz as such and so on.
Sounds reasonable...
adam777 wrote:
Now, my first attempt was trying to access the HW directly using mplayer's --ao=alsa:device=hw=0.0 and not limiting the format.
Both samples played at correct sample rate but were outputted at 16 bit, since the hardware supposedly do not support floating point, and the output reverted to default.
Note the Format floatle is not supported by hardware part.
This seemed a bit weird as using alsa without accessing the hardware directly (going through vmix), the output is indeed floating point.
So yes, your hardware doesn't support floating point. Don't compare the hw device to the more higher level interfaces (default,surround,plughw). The higher level interfaces provide additional sample formats which will be converted on the fly to the specific hw device capabilities.
adam777 wrote:Bottom line, is there a way to make everything sent to card as-is?
That would depend on the input format. Even some 24bit audio files may need some additional padding. If you want to know what your card supports, compile and run the following program: http://www.volkerschatz.com/noise/alsacap.c.
For example, this the output on my machine:
Card 0, ID `Intel', name `HDA Intel'
Device 0, ID `ALC268 Analog', name `ALC268 Analog', 1 subdevices (1 available)
2 channels, sampling rate 44100..192000 Hz
Sample formats: S16_LE, S32_LE
Subdevice 0, name `subdevice #0'
Last edited by GogglesGuy (2012-10-02 03:09:43)

ALSA: Recording Playback Audio with HDA-Intel

Hello,
My alsa-info. I only use analog output.
Alsamixer looks like this: http://ompldr.org/vYmE4Mw http://ompldr.org/vYmE4NA
I would like to record audio which is being played back on my computer. I need it for recording of gameplay videos. I want to use only ALSA for this and want to avoid PulseAudio, if possible.
Recording audio playing on the computer with Alsamixer tells me that I should use the "Mix" control of the Capture device. However, HDA-Intel appears not to provide such a control.
Does anyone know how I can create such control? Or perhaps I can somehow fake a capture device, that in fact simply plays back the hw:0,0 playback device?
Perhaps the snd-aloop module is a solution.
I will now start reading about .asoundrc files but wanted to leave this question here, because I feel that I'm likely to fail in my search for knowledge.
Cheers

I can't see a straightforward way to configure icecast to do this on the Arch Wiki.
I tried alsaloop, but it blocks devices when in use. Maybe I'm doing it wrong.
alsaloop -C hw:0,0 -P hw:0,0 -t 10000
I'm not the first to have this problem (thread of 2010-05-22)
I managed to get play-to-file working but ffmpeg doesn't seem to be able to handle the raw output and I'd have to make "default" play to file, as most games don't allow to set what alsa device to use for sound.

[SOLVED] alsa daemon error during init with Thinkpad X220

Hi,
I just updated my system (including a kernel update). After reboot, I found alsa daemon, which is used to restore previous volume level, failed to start. I tried to start that daemon manually after the boot process is completed, and that produced an error message
sudo rc.d start alsa
:: Restoring ALSA Levels [BUSY]
Found hardware: "HDA-Intel" "Intel CougarPoint HDMI" "HDA:14f1506e,17aa21da,00100000 HDA:80862805,80860101,00100000" "0x17aa" "0x21da"
Hardware is initialized using a generic method
[FAIL]
However, the volume level restored successfully. It seems that recent Intel plantform provides two soundcard, and that error only affected the secondary card, which has nothing to do about system volume level.
Anyway, I hate to have a ``[FAIL]'' During init process. Any suggesting to get rid of that problem? Thanks
System Info: (I use linux-ck as my kernel with modified config)
uname -a
Linux Thomas 3.2.1-2-ck #1 SMP PREEMPT Sun Jan 15 00:16:41 CST 2012 x86_64 Intel(R) Core(TM) i7-2620M CPU @ 2.70GHz GenuineIntel GNU/Linux
lspci
00:00.0 Host bridge: Intel Corporation 2nd Generation Core Processor Family DRAM Controller (rev 09)
00:02.0 VGA compatible controller: Intel Corporation 2nd Generation Core Processor Family Integrated Graphics Controller (rev 09)
00:16.0 Communication controller: Intel Corporation 6 Series/C200 Series Chipset Family MEI Controller #1 (rev 04)
00:19.0 Ethernet controller: Intel Corporation 82579LM Gigabit Network Connection (rev 04)
00:1a.0 USB controller: Intel Corporation 6 Series/C200 Series Chipset Family USB Enhanced Host Controller #2 (rev 04)
00:1b.0 Audio device: Intel Corporation 6 Series/C200 Series Chipset Family High Definition Audio Controller (rev 04)
00:1c.0 PCI bridge: Intel Corporation 6 Series/C200 Series Chipset Family PCI Express Root Port 1 (rev b4)
00:1c.1 PCI bridge: Intel Corporation 6 Series/C200 Series Chipset Family PCI Express Root Port 2 (rev b4)
00:1c.3 PCI bridge: Intel Corporation 6 Series/C200 Series Chipset Family PCI Express Root Port 4 (rev b4)
00:1c.4 PCI bridge: Intel Corporation 6 Series/C200 Series Chipset Family PCI Express Root Port 5 (rev b4)
00:1c.6 PCI bridge: Intel Corporation 6 Series/C200 Series Chipset Family PCI Express Root Port 7 (rev b4)
00:1d.0 USB controller: Intel Corporation 6 Series/C200 Series Chipset Family USB Enhanced Host Controller #1 (rev 04)
00:1f.0 ISA bridge: Intel Corporation QM67 Express Chipset Family LPC Controller (rev 04)
00:1f.2 SATA controller: Intel Corporation 6 Series/C200 Series Chipset Family 6 port SATA AHCI Controller (rev 04)
00:1f.3 SMBus: Intel Corporation 6 Series/C200 Series Chipset Family SMBus Controller (rev 04)
03:00.0 Network controller: Intel Corporation Centrino Advanced-N + WiMAX 6250 (rev 5e)
0d:00.0 System peripheral: Ricoh Co Ltd Device e823 (rev 07)
0e:00.0 USB controller: NEC Corporation uPD720200 USB 3.0 Host Controller (rev 04)
config.gz:
http://codepad.org/XtUSbtdw
(Edit typo)
Last edited by cap_sensitive (2012-01-17 06:12:01)

This morning, I performed a system update and I did not see any errors with alsa when I rebooted. Audio hardware is:
00:1b.0 Audio device: Intel Corporation 5 Series/3400 Series Chipset High Definition Audio (rev 05)
Have you tried manually restarting alsa after the system boots up to see if you get the same error? You can restart alsa with the following command:
sudo /etc/rc.d/alsa force-restart
Post your results. If the error occurs during the manual restart, it should also include some details about what caused the error

[SOLVED] No sound from speaker with alsa on lenovo g585

When playing any kind of sound, my headphones work, but the internal speakers are silent. In alsamixer both Headphone and Speaker are unmuted. So far I made a ~/.asoundrc file, so the HDMI device wont be the default.
~/.asoundrc
pcm.!default {
type hw
card Generic_1
ctl.!default {
type hw
card Generic_1
aplay -l
**** List of PLAYBACK Hardware Devices ****
card 0: Generic [HD-Audio Generic], device 3: HDMI 0 [HDMI 0]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 1: Generic_1 [HD-Audio Generic], device 0: CX20590 Analog [CX20590 Analog]
Subdevices: 1/1
Subdevice #0: subdevice #0
lspci -k
00:01.1 Audio device: Advanced Micro Devices, Inc. [AMD/ATI] Wrestler HDMI Audio [Radeon HD 6250/6310]
Subsystem: Lenovo Device 397f
Kernel driver in use: snd_hda_intel
Kernel modules: snd_hda_intel
00:14.2 Audio device: Advanced Micro Devices, Inc. [AMD] FCH Azalia Controller (rev 01)
Subsystem: Lenovo Device 397f
Kernel driver in use: snd_hda_intel
Kernel modules: snd_hda_intel
alsa-info.sh output is here.
After reading the wiki, and looking at tihs I tried setting the model in /etc/modprobe.d/modprobe.conf to laptop, and to ideapad, but none of them worked. Is it only the matter of getting the model right, or is something else wrong?
Last edited by matyilona200 (2013-12-22 23:07:57)

I found that Auto-Mute Mode was enabled and my headphones where plugged in all the time, disableing it solved the problem.
Last edited by matyilona200 (2013-12-22 23:09:09)

[Solved]Alsa funkiness (sound breaking)

(Note: I originally thought this was a steam issue, but turns out it's not only steam that can break my sound (see the update on the bottom))
I get an issue with Steam where I get a short (less than 1s) sound on repeat/loop (usually it's some part of the message notification sound, or a trailer that I was watching) sounding a bit like a broken record, randomly occurring after keeping steam open for long periods of time in the background. Only fix I've found so far is to restart steam.
I have the Enlightenmen 18 Window Manager.
Notable errors:
Fontconfig error: "/etc/fonts/conf.d/10-scale-bitmap-fonts.conf", line 70: non-double matrix element
Fontconfig warning: "/etc/fonts/conf.d/10-scale-bitmap-fonts.conf", line 78: saw unknown, expected number
[0706/170950:ERROR:object_proxy.cc(239)] Failed to call method: org.freedesktop.DBus.Error.ServiceUnknown: The name org.freedesktop.NetworkManager was not provided by any .service files
[0706/170950:WARNING:proxy_service.cc(958)] PAC support disabled because there is no system implementation
PulseAudio connect failed (used only for Mic Volume Control) with error: Access denied
** (steam:7341): WARNING **: Could not initialize NMClient /org/freedesktop/NetworkManager: The name org.freedesktop.NetworkManager was not provided by any .service files
(steam:7341): LIBDBUSMENU-GLIB-WARNING **: Trying to remove a child that doesn't believe we're it's parent.
[0706/171536:ERROR:reference_audio_renderer.cc(46)] Not implemented reached in virtual void media::ReferenceAudioRenderer::OnCreated(media::AudioOutputController*)
[0706/171536:ERROR:reference_audio_renderer.cc(50)] Not implemented reached in virtual void media::ReferenceAudioRenderer::OnPlaying(media::AudioOutputController*)
[0706/171657:ERROR:reference_audio_renderer.cc(54)] Not implemented reached in virtual void media::ReferenceAudioRenderer::OnPaused(media::AudioOutputController*)
AL lib: pulseaudio.c:612: Context did not connect: Access denied
I don't have any reliable method to reproduce the issue, and it does not seem to output an error of it's own in the terminal output when it occurs. I have noticed that this is more likely to occur when steam has been open for prolonged periods. I am not using pulseaudio, just Alsa. I also noticed when I was running a Wine game (not through steam) that this issue occurred, and it seemed to contaminate the sound outputs for whine too (I got like a "double" sound for every sound effect the game played). If the sound was completely broken on steam and making a loop like I explained at the start of this post and then I close steam while the Wine game is running, the sound from steam will remain until I shut down the game too (i.e. seems that this isn't just breakage internally in steam, but in alsa itself)
The one thing I have tried to solve this was install lib32-libpulse, this did not seem to fix the errors nor the problem. I imagine installing pulseaudio could fix this problem, but pulseaudio downright hates my sound card, so I would rather not use it.
I also have two other issues too that are less severe.
1: Mouse cursor sometimes gets locked in steam instance (Right clicking will show the steam context menu, pressing "stop" in that context menu will solve the issue. This sometimes happens when I use "Ctrl + C"(copy) or "Ctrl + V"(paste) when I'm typing in the discussions)
2: Fullscreen videos will not go fullscreen (they will only fill the steam window)
Update:
I found a bulletproof way to make a similar issue occur.
If I open facebook in firefox, and get a message notification while I am playing a movie in Cmplayer (my media player of choice) the sound in CMplayer breaks (but firefox's sound keeps working normally)
Update2:
I was rather desperate to get this fixed so I installed pulseaudio as much as I despise it, the quality of my audio is damaged by it, but this problem was as I suspected fixed by installing it. I am guessing pulseaudio is set up partially to prevent issues like this one, but I would relaly prefer not to have it. This does not rid me of all the errors however (reference_audio_renderer errors persist) but it does create a new error for steam:
[0706/215704:ERROR:alsa_output.cc(684)] Failed querying delay: Input/output error
Last edited by rabcor (2014-07-11 16:28:34)

I always had to have an asound.conf in the name of upmixing 2.0 to 5.1 This is the content of my asound.conf before I made the changes. I also had a problem with a game where I had crackling noises on 48khz so I had to tell alsa to force 96khz (or 44.1khz but since my card supports 96khz that's what I went with) to avoid that.
pcm.!default {
slave.pcm "surround51"
slave.channels 6
type route
# Front and rear
ttable.0.0 0.7
ttable.1.1 0.7
ttable.2.2 0.6
ttable.3.3 0.6
# Center and LFE
ttable.4.4 1
ttable.5.5 1
# Front left/right to center
ttable.0.4 0.5
ttable.1.4 0.5
# Front left/right to rear
ttable.0.2 0.5
ttable.1.3 0.5
Now it looks like this and everything seems to be working fine (except that mono sound only plays from my left speaker and I haven't defined a way to downmix 7.1 to 5.1):
pcm.dmixed {
type asym
playback.pcm {
# See plugin:dmix at http://www.alsa-project.org/alsa-doc/alsa-lib/pcm_plugins.html
type dmix
# Don't block other users, e.g. the Timidity midi-player daemon
# http://www.alsa-project.org/alsa-doc/alsa-lib/pcm_plugins.html
ipc_key_add_uid true
ipc_key 5678293
ipc_perm 0660
ipc_gid audio
# Don't put the rate here! Otherwise it resets the rate & channels set below, as shown by: cat /proc/asound/card0/pcm0p/sub0/hw_params
slave {
# 2 for stereo, 6 for surround51, 8 for surround71
channels 6
pcm {
# mplayer chooses S32_LE, but others usually S16_LE
format S32_LE
#format S16_LE
# 44100 or 48000
# 44100 for music, 48000 is compatible with most h/w
#rate 44100
rate 96000
# http://www.alsa-project.org/alsa-doc/alsa-lib/pcm_plugins.html
# Maybe helps
nonblock true
type hw
card 0
device 0
subdevice 0
# mplayer2 chooses 1024
# period_size 512 with buffer_size 16384 stops crackling in xmame
# 320 breaks flash - https://bbs.archlinux.org/viewtopic.php?id=129458
#period_size 512
period_size 1024
#period_size 512
# 4096 might make sound crackle
# mplayer2 chooses 8192. Half-Life 2 chooses 16384.
# If too large, use CONFIG_SND_HDA_PREALLOC_SIZE=2048
buffer_size 16384
capture.pcm "hw:0"
pcm.!default{
type plug
slave.pcm "upmix20_51"
pcm.!surround20 {
type plug
slave.pcm "upmix20_51"
pcm.!surround40 {
type plug
slave.pcm "dmixed"
route_policy duplicate
pcm.!surround51 {
type plug
slave.pcm "dmixed"
pcm.upmix20_51 {
slave.pcm "dmixed"
slave.channels 6
type route
# Front and rear
ttable.0.0 0.7
ttable.1.1 0.7
ttable.2.2 0.6
ttable.3.3 0.6
# Center and LFE
ttable.4.4 1
ttable.5.5 1
# Front left/right to center
ttable.0.4 0.5
ttable.1.4 0.5
# Front left/right to rear
ttable.0.2 0.5
ttable.1.3 0.5
Is there a better way for me to achieve the results of the above config? Because I'm all ears.
Last edited by rabcor (2014-07-08 11:09:23)

[SOLVED] ALSA firmware not loading properly with kernel 3.8.6-1

Hi All,
I have a E-MU 1820 card + breakout box that have been working beautiful under Archlinux for years but now something happen.
It still work OK under windows (dual boot same computer).
Normally during boot all indicators on the breakout box blink and then only thin show is clock setting indicator. But now all the indicators still blink but instead it is a -12 dB indicator that light up and not the clock settings and no sound.
dmesg show following.
[christer@Arch ~]$ dmesg | grep emu
[ 5.269612] emu1010: Special config.
[ 5.269731] emu1010: EMU_HANA_ID = 0x7f
[ 5.274685] emu1010: firmware file = emu/hana.fw, size = 0x133a4
[ 13.465727] emu1010: Hana Firmware loaded
[ 13.465776] emu1010: Hana version: 3.4
[ 13.465864] emu1010: Card options = 0x0
[ 13.465888] emu1010: Card options = 0x0
[ 13.466379] emu1010: Card options3 = 0x0
[ 14.486723] emu1010: Loading Audio Dock Firmware
[ 16.477785] emu1010: EMU_HANA+DOCK_IRQ_STATUS = 0x36
[ 16.477809] emu1010: EMU_HANA+DOCK_ID = 0x55
[ 16.477809] emu1010: Audio Dock Firmware loaded
[christer@Arch ~]$
aplay -l looks like it did before.
[christer@Arch ~]$ aplay -l
**** List of PLAYBACK Hardware Devices ****
card 0: EMU1010 [E-mu 1010 [MAEM8810]], device 0: emu10k1 [ADC Capture/Standard PCM Playback]
Subdevices: 32/32
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 #6: subdevice #6
Subdevice #7: subdevice #7
Subdevice #8: subdevice #8
Subdevice #9: subdevice #9
Subdevice #10: subdevice #10
Subdevice #11: subdevice #11
Subdevice #12: subdevice #12
Subdevice #13: subdevice #13
Subdevice #14: subdevice #14
Subdevice #15: subdevice #15
Subdevice #16: subdevice #16
Subdevice #17: subdevice #17
Subdevice #18: subdevice #18
Subdevice #19: subdevice #19
Subdevice #20: subdevice #20
Subdevice #21: subdevice #21
Subdevice #22: subdevice #22
Subdevice #23: subdevice #23
Subdevice #24: subdevice #24
Subdevice #25: subdevice #25
Subdevice #26: subdevice #26
Subdevice #27: subdevice #27
Subdevice #28: subdevice #28
Subdevice #29: subdevice #29
Subdevice #30: subdevice #30
Subdevice #31: subdevice #31
card 0: EMU1010 [E-mu 1010 [MAEM8810]], device 2: emu10k1 efx [Multichannel Capture/PT Playback]
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 #6: subdevice #6
Subdevice #7: subdevice #7
card 0: EMU1010 [E-mu 1010 [MAEM8810]], device 3: emu10k1 [Multichannel Playback]
Subdevices: 1/1
Subdevice #0: subdevice #0
[christer@Arch ~]$
lsmod
[christer@Arch ~]$ lsmod | grep emu
snd_emu10k1 140245 1
snd_util_mem 2403 1 snd_emu10k1
snd_hwdep 6428 1 snd_emu10k1
snd_ac97_codec 113456 1 snd_emu10k1
snd_rawmidi 18831 1 snd_emu10k1
snd_seq_device 5268 5 snd_seq,snd_rawmidi,snd_seq_oss,snd_emu10k1,snd_seq_dummy
snd_pcm 78146 3 snd_pcm_oss,snd_ac97_codec,snd_emu10k1
snd_page_alloc 7426 2 snd_pcm,snd_emu10k1
snd_timer 18934 3 snd_pcm,snd_seq,snd_emu10k1
snd 60156 13 snd_pcm_oss,snd_ac97_codec,snd_hwdep,snd_timer,snd_pcm,snd_seq,snd_rawmidi,snd_seq_oss,snd_emu10k1,snd_seq_device,snd_mixer_oss
[christer@Arch ~]$
From what I can see all looks OK, like it did when it worked, but something is not at it should.
Any thoughts or ideas about what might have happen or how to progress in troubleshooting.
All input most welcome.
I found info describing problem with firmware loading, but it is over my head.
http://git.kernel.org/cgit/linux/kernel … 54cf7008ab
Output from Aadebug ( http://alsa.opensrc.org/Aadebug )
[christer@Arch ~]$ ./alsadebug
ALSA Audio Debug v0.2.0 - Fri Apr 12 20:46:05 CEST 2013
http://alsa.opensrc.org/aadebug
http://www.gnu.org/licenses/agpl-3.0.txt
Kernel ----------------------------------------------------
Linux Arch 3.8.6-1-ARCH #1 SMP PREEMPT Sat Apr 6 07:27:01 CEST 2013 x86_64 GNU/Linux
Advanced Linux Sound Architecture Driver Version k3.8.6-1-ARCH.
Loaded Modules --------------------------------------------
snd_emu10k1 139271 2
snd_util_mem 2339 1 snd_emu10k1
snd_hwdep 6364 1 snd_emu10k1
snd_pcm_oss 38511 0
snd_mixer_oss 14995 1 snd_pcm_oss
snd_ac97_codec 112216 1 snd_emu10k1
snd_seq_dummy 1463 0
snd_seq_oss 29098 0
snd_seq_midi_event 5660 1 snd_seq_oss
snd_rawmidi 18742 1 snd_emu10k1
snd_seq 49946 5 snd_seq_midi_event,snd_seq_oss,snd_seq_dummy
snd_seq_device 5180 5 snd_seq,snd_rawmidi,snd_seq_oss,snd_emu10k1,snd_seq_dummy
snd_pcm 76956 4 snd_pcm_oss,snd_ac97_codec,snd_emu10k1
snd_page_alloc 7298 2 snd_pcm,snd_emu10k1
snd_timer 18687 3 snd_pcm,snd_seq,snd_emu10k1
snd 58893 14 snd_pcm_oss,snd_ac97_codec,snd_hwdep,snd_timer,snd_pcm,snd_seq,snd_rawmidi,snd_seq_oss,snd_emu10k1,snd_seq_device,snd_mixer_oss
Proc Asound -----------------------------------------------
0 [EMU1010 ]: Audigy2 - E-mu 1010 [MAEM8810]
E-mu 1010 [MAEM8810] (rev.3, serial:0x40011102) at 0xd000, irq 19
1: : sequencer
2: [ 0- 0]: hardware dependent
3: [ 0- 1]: raw midi
4: [ 0- 0]: raw midi
5: [ 0- 3]: digital audio playback
6: [ 0- 2]: digital audio playback
7: [ 0- 2]: digital audio capture
8: [ 0- 1]: digital audio capture
9: [ 0- 0]: digital audio playback
10: [ 0- 0]: digital audio capture
11: [ 0] : control
33: : timer
00-00: EMU10K1 (FX8010)
00-00: emu10k1 : ADC Capture/Standard PCM Playback : playback 32 : capture 1
00-01: emu10k1 mic : Mic Capture : capture 1
00-02: emu10k1 efx : Multichannel Capture/PT Playback : playback 8 : capture 1
00-03: emu10k1 : Multichannel Playback : playback 1
Client info
cur clients : 3
peak clients : 3
max clients : 192
Client 0 : "System" [Kernel]
Port 0 : "Timer" (Rwe-)
Port 1 : "Announce" (R-e-)
Connecting To: 15:0
Client 14 : "Midi Through" [Kernel]
Port 0 : "Midi Through Port-0" (RWe-)
Client 15 : "OSS sequencer" [Kernel]
Port 0 : "Receiver" (-we-)
Connected From: 0:1
Client 16 : "E-mu 1010 [MAEM8810]" [Kernel]
Port 0 : "Audigy MPU-401 (UART)" (RWeX)
Port 32 : "Audigy MPU-401 #2" (RWeX)
Client 17 : "Emu10k1 WaveTable" [Kernel]
Port 0 : "Emu10k1 Port 0" (-We-)
Port 1 : "Emu10k1 Port 1" (-We-)
Port 2 : "Emu10k1 Port 2" (-We-)
Port 3 : "Emu10k1 Port 3" (-We-)
Dev Snd ---------------------------------------------------
total 0
drwxr-xr-x 2 root root 60 Apr 12 20:42 by-path
crw-rw----+ 1 root audio 116, 11 Apr 12 20:42 controlC0
crw-rw----+ 1 root audio 116, 2 Apr 12 20:42 hwC0D0
crw-rw----+ 1 root audio 116, 12 Apr 12 20:46 hwC0D2
crw-rw----+ 1 root audio 116, 4 Apr 12 20:42 midiC0D0
crw-rw----+ 1 root audio 116, 3 Apr 12 20:42 midiC0D1
crw-rw----+ 1 root audio 116, 13 Apr 12 20:46 midiC0D2
crw-rw----+ 1 root audio 116, 14 Apr 12 20:46 midiC0D3
crw-rw----+ 1 root audio 116, 10 Apr 12 20:42 pcmC0D0c
crw-rw----+ 1 root audio 116, 9 Apr 12 20:43 pcmC0D0p
crw-rw----+ 1 root audio 116, 8 Apr 12 20:42 pcmC0D1c
crw-rw----+ 1 root audio 116, 7 Apr 12 20:42 pcmC0D2c
crw-rw----+ 1 root audio 116, 6 Apr 12 20:42 pcmC0D2p
crw-rw----+ 1 root audio 116, 5 Apr 12 20:42 pcmC0D3p
crw-rw----+ 1 root audio 116, 1 Apr 12 20:42 seq
crw-rw----+ 1 root audio 116, 33 Apr 12 20:42 timer
CPU -------------------------------------------------------
model name : Intel(R) Core(TM)2 Duo CPU E8400 @ 3.00GHz
cpu MHz : 2000.000
model name : Intel(R) Core(TM)2 Duo CPU E8400 @ 3.00GHz
cpu MHz : 2000.000
RAM -------------------------------------------------------
MemTotal: 4051696 kB
SwapTotal: 51196 kB
Hardware --------------------------------------------------
01:00.0 VGA compatible controller: NVIDIA Corporation G94 [GeForce 9600 GT] (rev a1)
05:01.0 Multimedia audio controller: Creative Labs SB Audigy (rev 03)
Interupts -------------------------------------------------
CPU0 CPU1
0: 48 0 IO-APIC-edge timer
1: 1 1 IO-APIC-edge i8042
6: 1 2 IO-APIC-edge floppy
8: 0 1 IO-APIC-edge rtc0
9: 0 0 IO-APIC-fasteoi acpi
12: 4 1 IO-APIC-edge i8042
16: 529 493 IO-APIC-fasteoi uhci_hcd:usb1, pata_jmicron, nvidia
18: 10654 10645 IO-APIC-fasteoi uhci_hcd:usb3, ehci_hcd:usb4, uhci_hcd:usb8, firewire_ohci, i801_smbus
19: 3621 3626 IO-APIC-fasteoi uhci_hcd:usb7, snd_emu10k1
21: 0 0 IO-APIC-fasteoi uhci_hcd:usb2
23: 0 0 IO-APIC-fasteoi ehci_hcd:usb5, uhci_hcd:usb6
44: 7637 7652 PCI-MSI-edge ahci
45: 11719 11736 PCI-MSI-edge eth0
NMI: 47 47 Non-maskable interrupts
LOC: 49551 45081 Local timer interrupts
SPU: 0 0 Spurious interrupts
PMI: 47 47 Performance monitoring interrupts
IWI: 0 0 IRQ work interrupts
RTR: 0 0 APIC ICR read retries
RES: 28730 28313 Rescheduling interrupts
CAL: 94 54 Function call interrupts
TLB: 3503 3509 TLB shootdowns
TRM: 0 0 Thermal event interrupts
THR: 0 0 Threshold APIC interrupts
MCE: 0 0 Machine check exceptions
MCP: 1 1 Machine check polls
ERR: 0
MIS: 0
[christer@Arch ~]$
output from systemctl show somw inacive sound related sevices.
alsa-restore.service loaded inactive dead Restore Sound Card State
alsa-store.service loaded inactive dead Store Sound Card State
auditd.service error inactive dead auditd.service
display-manager.service error inactive dead display-manager.service
dmeventd.service loaded inactive dead Device-mapper event daemon
emergency.service loaded inactive dead Emergency Shell
lvmetad.service loaded inactive dead LVM2 metadata daemon
plymouth-quit-wait.service error inactive dead plymouth-quit-wait.service
plymouth-start.service error inactive dead plymouth-start.service
rc-local-shutdown.service loaded inactive dead /etc/rc.local.shutdown Compatibility
rescue.service loaded inactive dead Rescue Shell
systemd-...console.service loaded inactive dead Dispatch Password Requests to Console
systemd-...rd-wall.service loaded inactive dead Forward Password Requests to Wall
systemd-fsck-root.service loaded inactive dead File System Check on Root Device
systemd-...e8ee844.service loaded inactive dead File System Check on /dev/disk/by-uuid/9751ee37-e769-4bbe-9436-2d8e5e8ee844
systemd-...ev-sda2.service loaded inactive dead File System Check on /dev/sda2
systemd-...ev-sda4.service loaded inactive dead File System Check on /dev/sda4
systemd-initctl.service loaded inactive dead /dev/initctl Compatibility Daemon
systemd-...l-flush.service loaded inactive dead Trigger Flushing of Journal to Persistent Storage
systemd-...ed-load.service loaded inactive dead Load Random Seed
systemd-...ed-save.service loaded inactive dead Save Random Seed
systemd-...ad-done.service loaded inactive dead Stop Read-Ahead Data Collection
systemd-shutdownd.service loaded inactive dead Delayed Shutdown Service
systemd-...s-clean.service loaded inactive dead Cleanup of Temporary Directories
systemd-...unlevel.service loaded inactive dead Update UTMP about System Runlevel Changes
systemd-...hutdown.service loaded inactive dead Update UTMP about System Shutdown
ypbind.service error inactive dead ypbind.service
emergency.target loaded inactive dead Emergency Mode
final.target loaded inactive dead Final Step
nss-lookup.target loaded inactive dead Host and Network Name Lookups
nss-user-lookup.target loaded inactive dead User and Group Name Lookups
remote-fs-pre.target loaded inactive dead Remote File Systems (Pre)
remote-fs-setup.target error inactive dead remote-fs-setup.target
rescue.target loaded inactive dead Rescue Mode
runlevel1.target error inactive dead runlevel1.target
runlevel2.target error inactive dead runlevel2.target
runlevel3.target error inactive dead runlevel3.target
runlevel4.target error inactive dead runlevel4.target
runlevel5.target error inactive dead runlevel5.target
shutdown.target loaded inactive dead Shutdown
syslog.target error inactive dead syslog.target
umount.target loaded inactive dead Unmount All Filesystems
All the best!
Christer
Last edited by agkbill (2013-04-15 18:41:03)

Thank you mich41!
It looks like a kernel bug.
I downgraded to 3.7.10-1 then firmware loaded OK and I could play sounds.
Upgraded to 3.8.6-1 then the firmware fail to load.
Something looks not ok with 3.8.6-1.
Best regards,
Christer

[SOLVED]Alsa: No sound in flash64, works in flash32

Hi Everyone,
So I had a devel of a time getting audio working on my computer, largely because I was an idiot. I got oss working fine, but then it wouldn't work with either chromium or firefox. With help from mich41 (https://bbs.archlinux.org/viewtopic.php?id=156936) I was able to get it working with just alsa, and I uninstalled everything related to oss, or at least I thought I did.
Now sound works everywhere except the 64bit flash plugin (extra/flashplugin 11.2.202.261-1) at /usr/lib/mozilla/plugins/libflashplayer.so. In either chromium or firefox, flash videos play with video, but no sound, I can get Google Play Music to work fine, but not video. I fixed this problem in firefox by installing aur/lib32-flashplugin 11.2.202.261-2. However, neither chromium nor google-chrome will use this by default.
Here is the key though:
When I play a flash video in chromium, I get the following error:
ALSA lib pcm_oss.c:397:(_snd_pcm_oss_open) Cannot open device /dev/dsp
The only references I can find to pcm_oss in the 64bit and 32bit versions of alsa-plugins:
alsa-plugins: /usr/lib/alsa-lib/libasound_module_pcm_oss.so
lib32-alsa-plugins: /usr/lib32/alsa-lib/libasound_module_pcm_oss.so
I have search for a while and I can't find a way to either disable oss completely, or to change the oss device, when oss isn't installed. (yaourt -Qs oss turns up nothing related to oss, and I haven't custom compiled anything)
The only thing I can think of that I don't really want to do is to remove flashplugin and symlink the 32 bit version to the 64 bit library location so that chromium is tricked into using the 32 bit one.
Or perhaps I should install oss and alsa side-by-side?? I am pretty sure that is impossible since oss detaches the alsa kernel modules.
Does anyone have any ideas? Does anyone else have this problem?
Also, the pepper version of flash for chromium also does not work.
Thanks!
Last edited by MikeDacre (2013-04-01 23:39:37)

I figured it out - pulseaudio was not set up properly.
I just reinstalled pulseaudio, configured it to use the analog output and not the HDMI, installed pulseaudio-alsa, and rebooted, and now everything works!

[solved]kernel 2.6.28.2-1 hda-intel

Here is the information about my device,
output from "lspci | grep -i aud > audinfo"
00:1b.0 Audio device: Intel Corporation 82801G (ICH7 Family) High Definition Audio Controller (rev 02)
I recently upgraded to kernel version 2.6.28.1-1 and then today 2.6.28.2-1 and in both versions sound is not working, it was working as prior to my 2.6.28.1-1 upgrade, My computer is a HP mini 1000.
I am wondering what I have to do, lsmod shows that the module snd_hda-intel is loaded,
output from "lsmod | grep -i snd > sndinfo"
snd_pcsp 11048 0
snd_seq_oss 31872 0
snd_seq_midi_event 8192 1 snd_seq_oss
snd_seq 49968 4 snd_seq_oss,snd_seq_midi_event
snd_seq_device 8204 2 snd_seq_oss,snd_seq
snd_hda_intel 412852 0
snd_hwdep 9092 1 snd_hda_intel
snd_pcm_oss 40192 0
snd_pcm 70020 3 snd_pcsp,snd_hda_intel,snd_pcm_oss
snd_timer 21384 2 snd_seq,snd_pcm
snd_page_alloc 10120 2 snd_hda_intel,snd_pcm
snd_mixer_oss 16512 1 snd_pcm_oss
snd 50852 10 snd_pcsp,snd_seq_oss,snd_seq,snd_seq_device,snd_hda_intel,snd_hwdep,snd_pcm_oss,snd_pcm,snd_timer,snd_mixer_oss
soundcore 8160 1 snd
I was thinking maybe I have to compile a module in or go digging around the kernel, But having semi-recently moved to Arch linux from Gentoo linux I have no idea howto do this in Arch, although I am unsure as to wheither or not I actualy have to.
I appologise if I have left anything out and/or made some dumb mistake.
And yes, all my channels in alsamixer are unmuted
Thanks again,
Ducky,
[SOLVED]
Solved by using kernel 2.6.29-3 and details can be found in this thread:
http://bbs.archlinux.org/viewtopic.php?pid=532903
Last edited by suicideducky (2009-04-14 22:54:00)

Don't worry, ask as many questions you like, that's how people learn new stuff
1. You can download the the latest kernel and build it with your own PKGBUILD or use an example found on wiki:
http://wiki.archlinux.org/index.php/Ker … n_with_ABS
2. Or you can use the official PKGBUILD via abs.
Install abs:
pacman -Sy abs
Specify abs to fetch only the linux kernel:
abs core/kernel26
Copy the kernel26 directory from /var/abs/core/kernel26 to your home directory. And then use makepkg to build your kernel. Further reference:
http://wiki.archlinux.org/index.php/ABS
It's important to know that if you build from abs, you'll have change the PKGBUILD, in order to patch the kernel that is being automatically downloaded and compiled. You need to instruct makepkg through PKGBUILD to include the new alsa driver. If you extract the kernel you downloaded, and know where to copy the new alsa files, you'll be able to add some lines to the PKGBUILD before it compiles the kernel.
3. Another option you have is to build the alsa module without compiling the entire kernel. I mean you have the kernel source installed, so all you need to do is ./configure, make and make install in the alsa-driver directory. Download the latest alsa driver and read the INSTALL file for further instructions.
Yea, the new version of alsa is 1.0.19, which will be officially included in the next kernel release.
Last edited by ahcaliskan (2009-02-18 11:35:59)

[SOLVED] Xmonad: trasparency doesn't work with compton intel

Dear All,
I am having a strange issue with Xmonad and I am not sure whether it is a problem of my Arch or Xmonad configuration. I decided to ask on Arch forum first, as my other computer running Arch has exactly the same Xmonad setup and it works fine. I would appreciate assistance from some Xmonad or Arch proficient users on this.
Basically, I have this piece of code in my xmonad.hs:
myLogHook :: Handle -> X ()
, logHook = myLogHook dzenLeftBar >> fadeInactiveLogHook 0.8
On the other computer this makes all the unfocused windowses slightly transparent. I find this feature very useful as it helps me focus on the window I am working with.
For some reason this doesn't work on my laptop. My laptop is using xf86-video-intel since the lspci gave me
%lspci | grep VGA
00:02.0 VGA compatible controller: Intel Corporation Mobile GM965/GL960 Integrated Graphics Controller (primary) (rev 03)
Also I am invoking compton-git from AUR but I have also tried with xcompmgr in my xinitrc and it gave exactly the same result - no transparency whatsoever.
I looked through my log file, but couldn't find anything relevant (please let me know if you can think of anything). How do I proceed with this issue? Do I try different composite managers? Do I try different drivers? Is there anyhting in X setup that I should include? Please let me know if you have any ideas.
Last edited by AlmostSurelyRob (2013-06-18 10:18:08)

I am very sorry. I've just discovered that neither compton nor xcompmgr were installed. It's not only solved, it should be marked as [NOT RAISED]. I was migrating my configuration and overlooked some erm... details.

Problem with hda-intel

Hi!
I'm posting a new topic because i'm sure anyone ever had this problem
I have a notebook amilo m1437 and i'm going to talk about the integrated subwoofer.. i noticed that it works only if the headphones are plugged in at boot.. seems that headphones work like a "switch" and turn on subwoofer
does anyone know how i can turn on subwoofer without this trick with headphones?!?! maybe settings something in /proc..
thanks for help in advance!!
Last edited by baghera (2007-08-26 18:56:55)

bump...

[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] ALSA no sound interrnal speaker with HDA Intel PCH

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