Hi everyone.
I have been struggling to force my built-in Realtek ALC288 sound card to play at any sampling rate higher than 48khz. The card is built into an Asus Mini-PC-PN40, and is capable of 192khz sampling rate. Under Windows, it plays out at 192khz, but under Linux it appears capped at 48khz.
I assume this is due to the Linux Alsa system using Intel HD default sound card drivers instead of card-specific drivers, but I have not found a way to circumvent this restriction.
An example of the issue would be:
Code:
cg@debian:~$ speaker-test c2 -r 192000
speaker-test 1.1.8
Playback device is default
Stream parameters are 192000Hz, S16_LE, 1 channels
Using 16 octaves of pink noise
Rate set to 192000Hz (requested 192000Hz)
Buffer size range from 192 to 2097152
Period size range from 64 to 699051
Using max buffer size 2097152
Periods = 4
was set period_size = 524288
was set buffer_size = 2097152
0 - Front Left
as you can see, the card "believes" it is playing at 192000 HZ.
Similarly, I can record a file at 192khz
Code:
cg@debian:~$ arecord -f S16_LE -r 192000 -d 30 testS16_LE.wav
Recording WAVE 'testS16_LE.wav' : Signed 16 bit Little Endian, Rate 192000 Hz, Mono
And the resulting file is 192khz, when checked with ffprobe.
The system "believes" it plays it out at 192khz:
Code:
cg@debian:~$ aplay testS16_LE.wav
Playing WAVE 'testS16_LE.wav' : Signed 16 bit Little Endian, Rate 192000 Hz, Mono
However, it does not - when I test the playout sampling rate at the time of playback, I get 48khz:
Code:
cg@debian:~$ cat /proc/asound/card0/pcm0p/sub0/hw_params
access: MMAP_INTERLEAVED
format: S16_LE
subformat: STD
channels: 2
rate: 48000 (48000/1)
period_size: 44100
buffer_size: 88200
Changing the Alsa default settings in the Alsa config files does not seem to have any effect.
Checking the sinks shows that the card is seen as 48khz only:
Code:
pacmd list-sinks
1 sink(s) available.
* index: 0
name: <alsa_output.pci-0000_00_0e.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: 9039
volume: front-left: 30129 / 46% / -20.25 dB, front-right: 30129 / 46% / -20.25 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 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: 0 <alsa_card.pci-0000_00_0e.0>
module: 6
properties:
alsa.resolution_bits = "16"
device.api = "alsa"
device.class = "sound"
alsa.class = "generic"
alsa.subclass = "generic-mix"
alsa.name = "ALC255 Analog"
alsa.id = "ALC255 Analog"
alsa.subdevice = "0"
alsa.subdevice_name = "subdevice #0"
alsa.device = "0"
alsa.card = "0"
alsa.card_name = "HDA Intel PCH"
alsa.long_card_name = "HDA Intel PCH at 0xa1210000 irq 128"
alsa.driver_name = "snd_hda_intel"
device.bus_path = "pci-0000:00:0e.0"
sysfs.path = "/devices/pci0000:00/0000:00:0e.0/sound/card0"
device.bus = "pci"
device.vendor.id = "8086"
device.vendor.name = "Intel Corporation"
device.product.id = "3198"
device.form_factor = "internal"
device.string = "front:0"
device.buffering.buffer_size = "352800"
device.buffering.fragment_size = "176400"
device.access_mode = "mmap+timer"
device.profile.name = "analog-stereo"
device.profile.description = "Analog Stereo"
device.description = "Built-in Audio Analog Stereo"
alsa.mixer_name = "Realtek ALC255"
alsa.components = "HDA:10ec0255,1043871d,00100002 HDA:8086280d,80860101,00100000"
module-udev-detect.discovered = "1"
device.icon_name = "audio-card-pci"
ports:
analog-output-headphones: Headphones (priority 9000, latency offset 0 usec, available: yes)
properties:
device.icon_name = "audio-headphones"
active port: <analog-output-headphones>
It appears, if I read this correctly. that Linux uses a generic HDA Intel PCH card driver for the Realtek ALC255.
I am running this under Debian but the same effect exists in Ubuntu.
Any ideas what I can do to increase the playback sampling rate?
Thanks in advance!
Christo
