No pulseaudio profiles for digital audio - Asus Xonar DGX

Hi all,
I am unable to usa the digital sound ports of my Asus Xonar DGX card with pulseaudio. I see four profiles for analogue audio in pavucontrol but none for digital SP/DIF.
ALSA recognizes the hardware:
$ aplay -l
**** List of PLAYBACK Hardware Devices ****
card 0: NVidia [HDA NVidia], device 3: HDMI 0 [HDMI 0]
card 1: DGX [Xonar DGX], device 0: Multichannel [Multichannel]
Subdevices: 0/1
Subdevice #0: subdevice #0
card 1: DGX [Xonar DGX], device 1: Digital [Digital]
Subdevices: 1/1
Subdevice #0: subdevice #0
and when using ALSA directly (e.g., by telling smplayer to use alsa device 1.1) digital audio works fine. However, pulseaudio apparently does not find device 1.1, I have only analogue profiles:
$ pacmd list-cards
2 card(s) available.
index: 0
index: 1
name: <alsa_card.pci-0000_06_04.0>
driver: <module-alsa-card.c>
owner module: 25
properties:
alsa.card = "1"
alsa.card_name = "Xonar DGX"
alsa.long_card_name = "C-Media Oxygen HD Audio at 0xd000, irq 19"
alsa.driver_name = "snd_oxygen"
device.bus_path = "pci-0000:06:04.0"
sysfs.path = "/devices/pci0000:00/0000:00:1c.7/0000:05:00.0/0000:06:04.0/sound/card1"
device.bus = "pci"
device.vendor.id = "13f6"
device.vendor.name = "C-Media Electronics Inc"
device.product.id = "8788"
device.product.name = "CMI8788 [Oxygen HD Audio]"
device.string = "1"
device.description = "CMI8788 [Oxygen HD Audio]"
module-udev-detect.discovered = "1"
device.icon_name = "audio-card-pci"
profiles:
input:analog-stereo: Analog Stereo Input (priority 60, available: unknown)
output:analog-stereo: Analog Stereo Output (priority 6000, available: unknown)
output:analog-stereo+input:analog-stereo: Analog Stereo Duplex (priority 6060, available: unknown)
off: Off (priority 0, available: unknown)
active profile: <output:analog-stereo+input:analog-stereo>
sinks:
alsa_output.pci-0000_06_04.0.analog-stereo/#1: CMI8788 [Oxygen HD Audio] Analog Stereo
sources:
alsa_output.pci-0000_06_04.0.analog-stereo.monitor/#1: Monitor of CMI8788 [Oxygen HD Audio] Analog Stereo
alsa_input.pci-0000_06_04.0.analog-stereo/#2: CMI8788 [Oxygen HD Audio] Analog Stereo
ports:
analog-input-front-mic: Front Microphone (priority 8500, latency offset 0 usec, available: unknown)
properties:
device.icon_name = "audio-input-microphone"
analog-input-mic: Microphone (priority 8700, latency offset 0 usec, available: unknown)
properties:
device.icon_name = "audio-input-microphone"
analog-input-linein: Line In (priority 8100, latency offset 0 usec, available: unknown)
properties:
analog-input-aux: Analog Input (priority 8000, latency offset 0 usec, available: unknown)
properties:
analog-output-headphones: Headphones (priority 9000, latency offset 0 usec, available: unknown)
properties:
device.icon_name = "audio-headphones"
Do you guys have any idea how to fix this?
Thanks in advance,
k0ol

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Last edited by 15goudreau (2014-08-04 21:56:36)

Search for vid and pid combination, ergo 13f6:8788 in your case. This should be one of the search terms, vid and pid could be also separated, other search terms could include oxygen, sonar, dgx and so on. Simple google search should turn something up.
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# Maintainer: Joakim Hernberg <[email protected]>
# Contributor: Ray Rashif <[email protected]>
# Contributor: timbosa <[email protected]>
# Contributor: Tobias Powalowski <[email protected]>
# Contributor: Thomas Baechler <[email protected]>
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md5sums=('b621207b3f6ecbb67db18b13258f8ea8'
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'a8126ad28c0a902a575397cacd099db2'
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# vim:set ts=8 sts=2 sw=2 et:
Then create the file '0007-asus-xonar-dgx.patch' with the following content:
--- a/sound/pci/oxygen/oxygen.c 2014-08-01 20:45:20.299191180 +0200
+++ b/sound/pci/oxygen/oxygen.c 2014-08-01 20:46:11.179947117 +0200
@@ -105,7 +105,7 @@
{ OXYGEN_PCI_SUBID(0x10b0, 0x0219), .driver_data = MODEL_CMEDIA_REF },
{ OXYGEN_PCI_SUBID(0x13f6, 0x0001), .driver_data = MODEL_CMEDIA_REF },
{ OXYGEN_PCI_SUBID(0x13f6, 0x0010), .driver_data = MODEL_CMEDIA_REF },
- { OXYGEN_PCI_SUBID(0x13f6, 0x8788), .driver_data = MODEL_CMEDIA_REF },
+ { OXYGEN_PCI_SUBID(0x13f6, 0x8788), .driver_data = MODEL_XONAR_DGX },
{ OXYGEN_PCI_SUBID(0x147a, 0xa017), .driver_data = MODEL_CMEDIA_REF },
{ OXYGEN_PCI_SUBID(0x1a58, 0x0910), .driver_data = MODEL_CMEDIA_REF },
/* Asus Xonar DG */
Finally run makepkg, should there be errors try removing the src dir first 'yes|rm -r src', then run makepkg again. Should there still be an error post here.
As for the patch, should there be no real difference there is also MODEL_XONAR_DG, might be worth trying out.
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I'm using MAM-A blank CD-R for digital audio and I would like to know which speed do I set to so I can get the best sound?Thanks Joe

When making new digital audio cd's which burn speed is best used with MAM-A 74min/650mb digital audio CD-R's?Thanks Joe

In theory, the slowest speed should result in the most accurate duplication (best sound). This will take much longer than if you were to use use faster speeds and whether that in practical terms is necessary is debatable. Remember, the quality of the disc will be no better than the source material.
Ciao.
PS: You may wish to experiment by making disks at different speeds and listen to them if can detect difference between them.

[SOLVED] No sound with Asus Xonar DGX

I've got no sound so I've looked into journalctl and there were some bluez and pulseaudio related erros. So i.a.w. this topic https://bbs.archlinux.org/viewtopic.php?id=155714 I did `systemctl enable bluetooth.service`. No more errors but still no sound.
Other things I did:
- unplug/plug headphones
- check whether anything is muted in alsamixer and the gnome sound applet (note: nothing is greyed-out, microphone picks up signals)
snd_oxygen i.e. ASUS XONAR is the sound card I would like to use and I hook my headphones to.
# cat /proc/asound/cards
0 [DGX ]: CMI8786 - Xonar DGX
C-Media Oxygen HD Audio at 0xee00, irq 17
1 [HDMI ]: HDA-Intel - HDA ATI HDMI
HDA ATI HDMI at 0xfd7fc000 irq 30
# lsmod | grep '^snd' | column -t
snd_hda_codec_hdmi 49263 1
snd_hda_intel 26387 2
snd_hda_controller 26857 1 snd_hda_intel
snd_oxygen 23783 2
snd_hda_codec 112621 3 snd_hda_codec_hdmi,snd_hda_intel,snd_hda_controller
snd_oxygen_lib 36709 1 snd_oxygen
snd_hwdep 17244 1 snd_hda_codec
snd_mpu401_uart 13507 1 snd_oxygen_lib
snd_rawmidi 26806 1 snd_mpu401_uart
snd_seq_device 13307 1 snd_rawmidi
snd_pcm 88779 5 snd_hda_codec_hdmi,snd_oxygen_lib,snd_hda_codec,snd_hda_intel,snd_hda_controller
snd_timer 26614 1 snd_pcm
snd 73436 19 snd_oxygen,snd_hwdep,snd_timer,snd_hda_codec_hdmi,snd_pcm,snd_rawmidi,snd_oxygen_lib,snd_hda_codec,snd_hda_intel,snd_mpu401_uart,snd_seq_device
# journalctl -b | grep -i bluez
Apr 26 12:13:52 arch1411 dbus[326]: [system] Activating via systemd: service name='org.bluez' unit='dbus-org.bluez.service'
Apr 26 12:13:52 arch1411 dbus[326]: [system] Successfully activated service 'org.bluez'
Apr 26 12:14:31 arch1411 /usr/lib/gdm/gdm-x-session[359]: Activating service name='org.bluez.obex'
Apr 26 12:14:31 arch1411 /usr/lib/gdm/gdm-x-session[359]: Successfully activated service 'org.bluez.obex'
[root@arch1411 hg1]# journalctl -b | grep -i audio
Apr 26 12:13:50 arch1411 kernel: snd_hda_intel 0000:01:00.1: Handle VGA-switcheroo audio client
Apr 26 12:13:52 arch1411 rtkit-daemon[432]: Successfully made thread 431 of process 431 (/usr/bin/pulseaudio) owned by '1000' high priority at nice level -11.
Apr 26 12:13:52 arch1411 rtkit-daemon[432]: Successfully made thread 456 of process 431 (/usr/bin/pulseaudio) owned by '1000' RT at priority 5.
Apr 26 12:13:52 arch1411 rtkit-daemon[432]: Successfully made thread 497 of process 431 (/usr/bin/pulseaudio) owned by '1000' RT at priority 5.
Apr 26 12:13:52 arch1411 rtkit-daemon[432]: Successfully made thread 498 of process 431 (/usr/bin/pulseaudio) owned by '1000' RT at priority 5.
Apr 26 15:37:00 arch1411 vlc.desktop[876]: [00007ffabc009ee8] mpgatofixed32 audio converter error: libmad error: bad main_data_begin pointer
Apr 26 15:37:02 arch1411 vlc.desktop[876]: [00007ffae0c244f8] mpgatofixed32 audio converter error: libmad error: bad main_data_begin pointer
# systemctl status pulseaudio.service
● pulseaudio.service
Loaded: not-found (Reason: No such file or directory)
Active: inactive (dead)
# systemctl start pulseaudio.service
Failed to start pulseaudio.service: Unit pulseaudio.service failed to load: No such file or directory.
Should I proceed per https://bbs.archlinux.org/viewtopic.php?id=149501 and remove pulseaudio daemon from the /etc/rc.conf file?
Last edited by bbarcher (2015-05-01 07:42:36)

systemctl --all --no-pager|grep -i pulse
returns nothing.
Thank you for your input, V1del.
$ pacmd list-cards
2 card(s) available.
index: 0
name: <alsa_card.pci-0000_01_00.1>
driver: <module-alsa-card.c>
owner module: 6
properties:
alsa.card = "0"
alsa.card_name = "HDA ATI HDMI"
alsa.long_card_name = "HDA ATI HDMI at 0xfd7fc000 irq 30"
alsa.driver_name = "snd_hda_intel"
device.bus_path = "pci-0000:01:00.1"
sysfs.path = "/devices/pci0000:00/0000:00:02.0/0000:01:00.1/sound/card0"
device.bus = "pci"
device.vendor.id = "1002"
device.vendor.name = "Advanced Micro Devices, Inc. [AMD/ATI]"
device.product.id = "aa88"
device.product.name = "Barts HDMI Audio [Radeon HD 6800 Series]"
device.string = "0"
device.description = "Barts HDMI Audio [Radeon HD 6800 Series]"
module-udev-detect.discovered = "1"
device.icon_name = "audio-card-pci"
profiles:
output:hdmi-stereo: Digital Stereo (HDMI) Output (priority 5400, available: unknown)
output:hdmi-surround: Digital Surround 5.1 (HDMI) Output (priority 300, available: unknown)
output:hdmi-surround71: Digital Surround 7.1 (HDMI) Output (priority 300, available: unknown)
off: Off (priority 0, available: unknown)
active profile: <output:hdmi-stereo>
sinks:
alsa_output.pci-0000_01_00.1.hdmi-stereo/#0: Barts HDMI Audio [Radeon HD 6800 Series] Digital Stereo (HDMI)
sources:
alsa_output.pci-0000_01_00.1.hdmi-stereo.monitor/#0: Monitor of Barts HDMI Audio [Radeon HD 6800 Series] Digital Stereo (HDMI)
ports:
hdmi-output-0: HDMI / DisplayPort (priority 5900, latency offset 0 usec, available: no)
properties:
device.icon_name = "video-display"
index: 1
name: <alsa_card.pci-0000_04_04.0>
driver: <module-alsa-card.c>
owner module: 24
properties:
alsa.card = "1"
alsa.card_name = "Xonar DGX"
alsa.long_card_name = "C-Media Oxygen HD Audio at 0xee00, irq 17"
alsa.driver_name = "snd_oxygen"
device.bus_path = "pci-0000:04:04.0"
sysfs.path = "/devices/pci0000:00/0000:00:05.0/0000:03:00.0/0000:04:04.0/sound/card1"
device.bus = "pci"
device.vendor.id = "13f6"
device.vendor.name = "C-Media Electronics Inc"
device.product.id = "8788"
device.product.name = "CMI8788 [Oxygen HD Audio]"
device.string = "1"
device.description = "CMI8788 [Oxygen HD Audio]"
module-udev-detect.discovered = "1"
device.icon_name = "audio-card-pci"
profiles:
input:analog-stereo: Analog Stereo Input (priority 60, available: unknown)
output:analog-stereo: Analog Stereo Output (priority 6000, available: unknown)
output:analog-stereo+input:analog-stereo: Analog Stereo Duplex (priority 6060, available: unknown)
off: Off (priority 0, available: unknown)
active profile: <output:analog-stereo+input:analog-stereo>
sinks:
alsa_output.pci-0000_04_04.0.analog-stereo/#1: CMI8788 [Oxygen HD Audio] Analog Stereo
sources:
alsa_output.pci-0000_04_04.0.analog-stereo.monitor/#1: Monitor of CMI8788 [Oxygen HD Audio] Analog Stereo
alsa_input.pci-0000_04_04.0.analog-stereo/#2: CMI8788 [Oxygen HD Audio] Analog Stereo
ports:
analog-input-front-mic: Front Microphone (priority 8500, latency offset 0 usec, available: unknown)
properties:
device.icon_name = "audio-input-microphone"
analog-input-mic: Microphone (priority 8700, latency offset 0 usec, available: unknown)
properties:
device.icon_name = "audio-input-microphone"
analog-input-linein: Line In (priority 8100, latency offset 0 usec, available: unknown)
properties:
analog-input-aux: Analog Input (priority 8000, latency offset 0 usec, available: unknown)
properties:
analog-output-headphones: Headphones (priority 9000, latency offset 0 usec, available: unknown)
properties:
device.icon_name = "audio-headphones"
$ pacmd list-sinks
2 sink(s) available.
index: 0
name: <alsa_output.pci-0000_01_00.1.hdmi-stereo>
driver: <module-alsa-card.c>
flags: HARDWARE DECIBEL_VOLUME LATENCY FLAT_VOLUME DYNAMIC_LATENCY
state: SUSPENDED
suspend cause: IDLE
priority: 9050
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 request: 0 KiB
max rewind: 0 KiB
monitor source: 0
sample spec: s16le 2ch 44100Hz
channel map: front-left,front-right
Stereo
used by: 0
linked by: 0
configured latency: 0.00 ms; range is 0.50 .. 1999.82 ms
card: 0 <alsa_card.pci-0000_01_00.1>
module: 6
properties:
alsa.resolution_bits = "16"
device.api = "alsa"
device.class = "sound"
alsa.class = "generic"
alsa.subclass = "generic-mix"
alsa.name = "HDMI 0"
alsa.id = "HDMI 0"
alsa.subdevice = "0"
alsa.subdevice_name = "subdevice #0"
alsa.device = "3"
alsa.card = "0"
alsa.card_name = "HDA ATI HDMI"
alsa.long_card_name = "HDA ATI HDMI at 0xfd7fc000 irq 30"
alsa.driver_name = "snd_hda_intel"
device.bus_path = "pci-0000:01:00.1"
sysfs.path = "/devices/pci0000:00/0000:00:02.0/0000:01:00.1/sound/card0"
device.bus = "pci"
device.vendor.id = "1002"
device.vendor.name = "Advanced Micro Devices, Inc. [AMD/ATI]"
device.product.id = "aa88"
device.product.name = "Barts HDMI Audio [Radeon HD 6800 Series]"
device.string = "hdmi:0"
device.buffering.buffer_size = "352768"
device.buffering.fragment_size = "176384"
device.access_mode = "mmap+timer"
device.profile.name = "hdmi-stereo"
device.profile.description = "Digital Stereo (HDMI)"
device.description = "Barts HDMI Audio [Radeon HD 6800 Series] Digital Stereo (HDMI)"
alsa.mixer_name = "ATI R6xx HDMI"
alsa.components = "HDA:1002aa01,00aa0100,00100200"
module-udev-detect.discovered = "1"
device.icon_name = "audio-card-pci"
ports:
hdmi-output-0: HDMI / DisplayPort (priority 5900, latency offset 0 usec, available: no)
properties:
device.icon_name = "video-display"
active port: <hdmi-output-0>
* index: 1
name: <alsa_output.pci-0000_04_04.0.analog-stereo>
driver: <module-alsa-card.c>
flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY FLAT_VOLUME DYNAMIC_LATENCY
state: SUSPENDED
suspend cause: IDLE
priority: 9059
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 request: 0 KiB
max rewind: 0 KiB
monitor source: 1
sample spec: s16le 2ch 48000Hz
channel map: front-left,front-right
Stereo
used by: 0
linked by: 0
configured latency: 0.00 ms; range is 0.50 .. 1837.50 ms
card: 1 <alsa_card.pci-0000_04_04.0>
module: 24
properties:
alsa.resolution_bits = "16"
device.api = "alsa"
device.class = "sound"
alsa.class = "generic"
alsa.subclass = "generic-mix"
alsa.name = "Multichannel"
alsa.id = "Multichannel"
alsa.subdevice = "0"
alsa.subdevice_name = "subdevice #0"
alsa.device = "0"
alsa.card = "1"
alsa.card_name = "Xonar DGX"
alsa.long_card_name = "C-Media Oxygen HD Audio at 0xee00, irq 17"
alsa.driver_name = "snd_oxygen"
device.bus_path = "pci-0000:04:04.0"
sysfs.path = "/devices/pci0000:00/0000:00:05.0/0000:03:00.0/0000:04:04.0/sound/card1"
device.bus = "pci"
device.vendor.id = "13f6"
device.vendor.name = "C-Media Electronics Inc"
device.product.id = "8788"
device.product.name = "CMI8788 [Oxygen HD Audio]"
device.string = "hw:1"
device.buffering.buffer_size = "352800"
device.buffering.fragment_size = "352800"
device.access_mode = "mmap+timer"
device.profile.name = "analog-stereo"
device.profile.description = "Analog Stereo"
device.description = "CMI8788 [Oxygen HD Audio] Analog Stereo"
alsa.mixer_name = "CMI8786"
alsa.components = "CS4245 CMI8786"
module-udev-detect.discovered = "1"
device.icon_name = "audio-card-pci"
ports:
analog-output-headphones: Headphones (priority 9000, latency offset 0 usec, available: unknown)
properties:
device.icon_name = "audio-headphones"
active port: <analog-output-headphones>
After all of the recommended solutions from the wiki I still get
$ systemctl status pulseaudio.service
● pulseaudio.service
Loaded: not-found (Reason: No such file or directory)
Active: inactive (dead)
$ pulseaudio -vvvv
I: [pulseaudio] main.c: setrlimit(RLIMIT_NICE, (31, 31)) failed: Operation not permitted
I: [pulseaudio] main.c: setrlimit(RLIMIT_RTPRIO, (9, 9)) failed: Operation not permitted
D: [pulseaudio] core-rtclock.c: Timer slack is set to 50 us.
D: [pulseaudio] core-util.c: RealtimeKit worked.
I: [pulseaudio] core-util.c: Successfully gained nice level -11.
I: [pulseaudio] main.c: This is PulseAudio 6.0
D: [pulseaudio] main.c: Compilation host: x86_64-unknown-linux-gnu
D: [pulseaudio] main.c: Compilation CFLAGS: -march=x86-64 -mtune=generic -O2 -pipe -fstack-protector-strong --param=ssp-buffer-size=4 -Wall -W -Wextra -Wno-long-long -Wno-overlength-strings -Wunsafe-loop-optimizations -Wundef -Wformat=2 -Wlogical-op -Wsign-compare -Wformat-security -Wmissing-include-dirs -Wformat-nonliteral -Wpointer-arith -Winit-self -Wdeclaration-after-statement -Wfloat-equal -Wmissing-prototypes -Wredundant-decls -Wmissing-declarations -Wmissing-noreturn -Wshadow -Wendif-labels -Wcast-align -Wstrict-aliasing -Wwrite-strings -Wno-unused-parameter -ffast-math -fno-common -fdiagnostics-show-option -fdiagnostics-color=auto
D: [pulseaudio] main.c: Running on host: Linux x86_64 3.19.3-3-ARCH #1 SMP PREEMPT Wed Apr 8 14:10:00 CEST 2015
D: [pulseaudio] main.c: Found 6 CPUs.
I: [pulseaudio] main.c: Page size is 4096 bytes
D: [pulseaudio] main.c: Compiled with Valgrind support: no
D: [pulseaudio] main.c: Running in valgrind mode: no
D: [pulseaudio] main.c: Running in VM: no
D: [pulseaudio] main.c: Optimized build: yes
D: [pulseaudio] main.c: FASTPATH defined, only fast path asserts disabled.
I: [pulseaudio] main.c: Machine ID is d36c9fa4b78f4c8cae7f7ad7897683e3.
I: [pulseaudio] main.c: Session ID is c1.
I: [pulseaudio] main.c: Using runtime directory /run/user/1000/pulse.
I: [pulseaudio] main.c: Using state directory /home/hg1/.config/pulse.
I: [pulseaudio] main.c: Using modules directory /usr/lib/pulse-6.0/modules.
I: [pulseaudio] main.c: Running in system mode: no
E: [pulseaudio] pid.c: Daemon already running.
E: [pulseaudio] main.c: pa_pid_file_create() failed.
Last edited by bbarcher (2015-04-27 19:16:41)

Digital Audio output for W series - Alternatives

There's a thread talking about the displayport does not have audio-output. Its true and some people get victimized by ill-trained sales people who think their displayport is standard. There's no literature on Lenovo website warning users about the audio issue either - such thread is closed with the fact that (some) other manufacturers in the same period does have have audio either and Lenovo think they should be in the same boat.
Let's not go into the blamish again, but discuss what we can do with existing options with alternative digital audio out and line-in that the W series really really missed.
We have (1) mini-dock and (2) USB soundcard listed as options
I currently use a USB DAC (digital to analog) system and it do really good in terms of stereo music enjoyment. However my particular system does not have digital manupilated sound like 3-D surround or multi-channel output and not suprisely, laptop USB generally produce greater latency than desktop USB ports.
Besides USB, we have 1394 and express card 34 as expansion options.
What's your setup for digital audio out? Anyone know any internal soundcard that fits the slots of Wireless USB/Turbo memory PCI ports that all W-series have?

Other alternatives include
I know on mackbooks I've seen firewire used as an input for a Weiss Dac if you're into two channel audio. Do any reiceivers take firewire as an input for hometheater? Has anyone tested Firewire outputs on their W series? I'm told that firewire is sometimes preferable to USB, I'm not sure why this would be, can USB support passthrough of DTS encoded signals?
In the case that firewire works and you Do not have a firewire source, maybe something like this would be of value It converts firewire to digital coaxial s/pdif and allows passthrough of DTS and AC3.
http://www.m-audio.com/images/global/manuals/FireWire-Audiophile_Manual.pdf
As I'm into audio the critical thing here (and what you're paying for when you purchase blue-ray, is maintaining the digital information from the disk in an unconverted form. You want a direct bit pass through, and you want as little jitter introduced to the signal as possible. You most certainly don't want the sound card getting in the way, Digtital to Analog converting and then reverse packaging it up and putting it out over the soundcard digital out. My problem is I'm not familiar enough with computer audio to know how exactly the sounds is transmitted from your blue ray drive to the digital interface you've chosen.
Anyone know why USB digital out would not be a prefered method?
Moderator note; comment removed which contravened forum rules

HT4806 Mac mini digital audio out through the headphone jack

I'm misled by this sentence for audio out port from the support page: "For digital audio, you can use Apple iPhone headset with microphone".
http://support.apple.com/kb/ht5539
It is there for all mac minis starting from 2010 model.
Does it mean electrical digital signal? Any chance to get S/PDIF coaxial signal?

If an optical signal is OK, you just need one of these:
http://www.amazon.com/dp/B00016W6Y6
To convert the optical signal to a co-ax one will need a more complex/expensive convertor.

Digital audio cables

I've been looking for digital audio cables and have noticed a wide price range, anywhere from $12.95 to $60.00. Why the wide variance? Is there that much difference in their capabilities, and if so, what are the differences?

In modern digital audio signals, as I stated earlier, either the signal reaches it's destination, or it does not. With error correction, even small dropouts do not cause signal loss. Since digital signals are not prone to the problems that plague analog signals (inductance, crosstalk, jitter, etc), then expensive cables are not needed. All that is needed is a cable of sufficient length and correct impedance. On consumer grade equipment, the difference between a $3 dollar cable and a $150 dollar cable are completely inaudible to the human ear. Since most audio signals come from a compressed source (mp2, wma, wav, etc), the audible differences are due to the quality o the DACs used, and not the cable that connects the source. Even on the few pieces of equipment that will play the uncompressed audio of a blu-ray disc, cables are much less critical than the equipment used to decode them. There is simply no reason for anyone to spend $50, $60 or more dollars for any digital cable. Extra "shielding" is not needed. Any of the other claims made by cable makers justifying their exorbitant prices are simply marketing hoopla, smoke and mirrors, and in some cases complete and utter nonsense. Any cable of appropriate length and impedance is sufficient.
If you find my post useful or informative, please click the icon below with the plus sign and star to give kudos. Thank you!

Mac Mini Digital Audio Output

Hi,
My friend is trying to use the only audio output from the Mac Mini (1.42 GHz) as a digital out. He was under the impression that when you hook up the right cable (which has bought and we double checked-optical audio cable with Toslink adaptor.)
But when we go to pref. menu there is no selection for digital audio out except for the regular line out.
Any ideas or tricks to getting digital out on this Mini?
Thanks,
Babak

Does he have the 1.42ghz G4 Mini? If so, that model is not a digital/optical audio out jack, at least not the 1.42ghz model I have. Did not check to see if a later update prior to the intel switch added the port to that model. My 1.66Ghz intel mini has the digital/optical audio out. Tell your firend to check the manual that came with his mini, it will tell you if it is a digital/optical out port or not.

Digital Audio Amplifier Testing

Hello my beloved member,
First of all I just to thank everyone for helping other members like me who are new in testing system.
I want to develop a tester for digital audio amplifier, i dont know where to start and what to do. Can someone tell me what hardware should I use, the software to use for it and some codes to start about the development of this tester.
Basically I will test a PCB of a digital audio amplifier that have a pogo pins that will have to contact the test points on the PCB and a connection going into the PC.
Any help will be greatly appreciated.
Electronics Manufacturing Tester Forum

Hello smoken,
I agree that a few details (or list of tests) would help us better help you. However, I think I can answer the software question right off the bat: LabVIEW of course! If you've never used LabVIEW before, there are plenty of resources to help you get started:
LabVIEW Guided Tour (Flash)
LabVIEW Introduction Course (Six Hours)
LabVIEW Introduction Course (Three Hours)
I suspect for the hardware you'll want to use some sort of DMM or DAQ board, though it really depends on the accuracy you want (in addition to the types of measurements). You might want to check out our customer solutions page to see if that sparks any ideas:
http://www.ni.com/solutions/customer_solutions.htm
I searched for "digial amplifier" and found quite a few articles. You can also contact your local technical representative to discuss options. (888-280-7645) In any case, let us know some specifics and I'm sure you'll find answers!
Hope this helps,
Charlie S.
Visit ni.com/gettingstarted for step-by-step help in setting up your system

[SOLVED] Xonar DGX

When the 3.14 kernel came out my sound stop working so I've been sticking to the 3.10 LTS release. Now that 3.14 is the LTS release, my audio is no longer working again. I am using the ASUS Xonar DGX sound card after my motherboard's sound card broke. I have tried both ALSA and PulseAudio and every kernel release up to the current 3.15-6 to no avail. Both the system and ALSA detect the sound card so I compiled my own kernel to make sure the module is enable which also did not work. Would this mean I have to stick to 3.13 forever (or until I get a new desktop)?
lspci | grep media
04:04.0 Multimedia audio controller: C-Media Electronics Inc CMI8788 [Oxygen HD Audio]
Last edited by Name Taken (2014-07-20 14:34:57)

emeres wrote:
Screenshots are insufficient for troubleshooting. Please post:
aplay -lL;
lspci -nn | grep -i audio;
lsmod | grep ^snd;
systool -v -m snd_oxygen;
amixer;
As far as I recall, CMI8788 Oxygen uses snd-oxygen, should this have changed correct the systool command.
null
Discard all samples (playback) or generate zero samples (capture)
default:CARD=DGX
Xonar DGX, Multichannel
Default Audio Device
sysdefault:CARD=DGX
Xonar DGX, Multichannel
Default Audio Device
front:CARD=DGX,DEV=0
Xonar DGX, Multichannel
Front speakers
surround21:CARD=DGX,DEV=0
Xonar DGX, Multichannel
2.1 Surround output to Front and Subwoofer speakers
surround40:CARD=DGX,DEV=0
Xonar DGX, Multichannel
4.0 Surround output to Front and Rear speakers
surround41:CARD=DGX,DEV=0
Xonar DGX, Multichannel
4.1 Surround output to Front, Rear and Subwoofer speakers
surround50:CARD=DGX,DEV=0
Xonar DGX, Multichannel
5.0 Surround output to Front, Center and Rear speakers
surround51:CARD=DGX,DEV=0
Xonar DGX, Multichannel
5.1 Surround output to Front, Center, Rear and Subwoofer speakers
surround71:CARD=DGX,DEV=0
Xonar DGX, Multichannel
7.1 Surround output to Front, Center, Side, Rear and Woofer speakers
iec958:CARD=DGX,DEV=0
Xonar DGX, Multichannel
IEC958 (S/PDIF) Digital Audio Output
**** List of PLAYBACK Hardware Devices ****
card 0: DGX [Xonar DGX], device 0: Multichannel [Multichannel]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 0: DGX [Xonar DGX], device 1: Digital [Digital]
Subdevices: 1/1
Subdevice #0: subdevice #0
04:04.0 Multimedia audio controller [0401]: C-Media Electronics Inc CMI8788 [Oxygen HD Audio] [13f6:8788]
snd_oxygen 16679 0
snd_oxygen_lib 30333 1 snd_oxygen
snd_mpu401_uart 5979 1 snd_oxygen_lib
snd_rawmidi 20278 1 snd_mpu401_uart
snd_seq_device 5244 1 snd_rawmidi
snd_pcm 83207 1 snd_oxygen_lib
snd_timer 19294 1 snd_pcm
snd 61276 7 snd_oxygen,snd_timer,snd_pcm,snd_rawmidi,snd_oxygen_lib,snd_mpu401_uart,snd_seq_device
Module = "snd_oxygen"
Attributes:
coresize = "16679"
initsize = "0"
initstate = "live"
refcnt = "0"
taint = ""
uevent = <store method only>
Parameters:
enable = "Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y"
id = "(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null)"
index = "-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1"
Sections:
.bss = "0xffffffffa05c7300"
.data = "0xffffffffa05c6e00"
.exit.text = "0xffffffffa05c5d55"
.gnu.linkonce.this_module= "0xffffffffa05c7080"
.init.text = "0xffffffffa05ca000"
.note.gnu.build-id = "0xffffffffa05c5d68"
.rodata = "0xffffffffa05c5dc0"
.rodata.str1.1 = "0xffffffffa05c67b0"
.rodata.str1.8 = "0xffffffffa05c6db8"
.strtab = "0xffffffffa05cb2e0"
.symtab = "0xffffffffa05ca020"
.text = "0xffffffffa05c4000"
__mcount_loc = "0xffffffffa05c6b40"
__param = "0xffffffffa05c6ae0"
Simple mixer control 'Headphone',0
Capabilities: pvolume pswitch pswitch-joined
Playback channels: Front Left - Front Right
Limits: Playback 0 - 255
Mono:
Front Left: Playback 255 [100%] [0.00dB] [on]
Front Right: Playback 255 [100%] [0.00dB] [on]
Simple mixer control 'Front Mic',0
Capabilities: cvolume cswitch cswitch-joined cswitch-exclusive
Capture exclusive group: 0
Capture channels: Front Left - Front Right
Limits: Capture -24 - 24
Front Left: Capture 0 [50%] [0.00dB] [off]
Front Right: Capture 0 [50%] [0.00dB] [off]
Simple mixer control 'Line',0
Capabilities: cvolume cswitch cswitch-joined cswitch-exclusive
Capture exclusive group: 0
Capture channels: Front Left - Front Right
Limits: Capture -24 - 24
Front Left: Capture 0 [50%] [0.00dB] [off]
Front Right: Capture 0 [50%] [0.00dB] [off]
Simple mixer control 'Mic',0
Capabilities: cvolume cswitch cswitch-joined cswitch-exclusive
Capture exclusive group: 0
Capture channels: Front Left - Front Right
Limits: Capture -24 - 24
Front Left: Capture 0 [50%] [0.00dB] [on]
Front Right: Capture 0 [50%] [0.00dB] [on]
Simple mixer control 'IEC958',0
Capabilities: pswitch pswitch-joined
Playback channels: Mono
Mono: Playback [on]
Simple mixer control 'IEC958 Loopback',0
Capabilities: pswitch pswitch-joined
Playback channels: Mono
Mono: Playback [off]
Simple mixer control 'IEC958 Validity Check',0
Capabilities: cswitch cswitch-joined
Capture channels: Mono
Mono: Capture [off]
Simple mixer control 'Aux',0
Capabilities: cvolume cswitch cswitch-joined cswitch-exclusive
Capture exclusive group: 0
Capture channels: Front Left - Front Right
Limits: Capture -24 - 24
Front Left: Capture 24 [100%] [12.00dB] [off]
Front Right: Capture 24 [100%] [12.00dB] [off]
Simple mixer control 'ADC High-pass Filter',0
Capabilities: cenum
Items: 'Active' 'Frozen'
Item0: 'Active'
Simple mixer control 'Analog Input Monitor',0
Capabilities: pvolume pvolume-joined pswitch pswitch-joined
Playback channels: Mono
Limits: Playback 0 - 1
Mono: Playback 1 [100%] [0.00dB] [off]
Simple mixer control 'Analog Output',0
Capabilities: penum
Items: 'Stereo Headphones' 'Stereo Headphones FP' 'Multichannel'
Item0: 'Stereo Headphones FP'
Simple mixer control 'Digital Input Monitor',0
Capabilities: pvolume pvolume-joined pswitch pswitch-joined
Playback channels: Mono
Limits: Playback 0 - 1
Mono: Playback 1 [100%] [0.00dB] [off]
Simple mixer control 'Stereo Upmixing',0
Capabilities: enum
Items: 'Front' 'Front+Surround'
Item0: 'Front+Surround'

HD audio playback on Asus Xonar DX (AV200 chip)

I just installed my new Asus Xonar DX (AV200 chip). First thing I'd like to try is playing back HD audio. But when I use mplayer to do so, it says this:
==========================================================================
Opening audio decoder: [ffmpeg] FFmpeg/libavcodec audio decoders
libavcodec version 54.71.100 (internal)
AUDIO: 96000 Hz, 2 ch, s32le, 0.1 kbit/0.00% (ratio: 16->768000)
Selected audio codec: [ffflac] afm: ffmpeg (FFmpeg FLAC audio)
==========================================================================
[AO OSS] audio_setup: Can't open audio device /dev/dsp: No such file or directory
AO: [alsa] 48000Hz 2ch s32le (4 bytes per sample)
Video: no video
Starting playback...
So why doesn't it play at 96000 Hz?
I use Alsa, no dmix, pulse or anything.
Last edited by halfgaar (2013-05-15 18:51:00)

Hmm, dmix is on by default, it would seem. I thought I didn't have dmix because:
I never needed to enable it before because I had an Audigy 2 card, which does internal resampling and hardware mixing, and I just plugged in my new Xonar DX.
I don't have anything about it in my asound.conf(s) or mplayer.conf
I couldn't play mplayer and audacious at the same time.
However:
I just found out that I can play two mplayers at the same time, so mplayer does use dmix, without me telling it to.
setting the dmix.rate actually does change at what rate mplayer plays.
This is weird to me.
The only thing I have in asound.conf are some jack things (which I don't use, actually):
ctl.jack40 {
type hw
card 0
pcm.jack40 {
# "asym" allows for different
# handling of in/out devices
type asym
playback.pcm {
# route for mmap workaround
type route
slave.pcm surround40
slave.channels 4
ttable.0.0 1
ttable.1.1 1
ttable.2.2 1
ttable.3.3 1
capture.pcm {
# 2 channels only
type hw
card 0
I want to make absolutely sure there is no resampling going on with any sound source (not just mplayer). So, either I don't have dmix at all and I have to accept that I can only play one sound source at a time, or dmix (or maybe Pulse?) should be smart and only resample when it's necessary.
How about those resampling algorithms? Are they smart enough not to resample when it's not necessary? Everybody always bashes Creative cards for always resampling, but now that we have cards that don't do it, and therefore lack the ability to hardware-mix, it would defeat the purpose if then the software does it all the time anyway.
edit: I found this here:
NOTE: For ALSA 1.0.9rc2 and higher you don't need to setup dmix for analogue output. Dmix is enabled by default for soundcards which don't support hardware mixing. You still need to set it up for digital outputs.
I guess it is on, but not fully, since mplayer can't play when I play music with Audacious.
And I think it's bizar it resamples to 48k, since it's a 192k card.
Last edited by halfgaar (2013-05-15 22:02:56)

IOS 6.1.4 released for iPhone 5 Adds an Updated Audio Profile for Speakerphone

Apple has just released an Over the Air update for the iPhone 5. It seems to only provide an updated audio profile for the speaker phone. Nothing else listed in the change log.
I would assume that there will be an iTunes downloadable update available shortly.

It obviously is because you didnt understand. The iphone 4's last update to IOS 6.1.3 in some cases caused the phones speakers to not work (in which this happened to mine and a few people I know) which some people got rid of their phones. HENCE the AUDIO UPDATE that they recently posted but only iPhone 5 can download the update. That is my problem. I have alot of apple products diesel vdub and have had apple for a long time but I can't seem to understand why they would do this and I posted this in hopes to find more people that have had this issue and if they did anything to help fix it this issue. Thank you for your concern though.

No pulseaudio profiles for digital audio - Asus Xonar DGX

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