hi,
i have an asus p5q-pro motherboard with ALC1200 chip on board and i'd like
to play a flac (24-bit/96kHz) samples to test the new amplifier.
afaics in /proc/asound/card0/codec\#0 it supports this pcm format.
Codec: Realtek ALC1200
Address: 0
AFG Function Id: 0x1 (unsol 1)
Vendor Id: 0x10ec0888
Subsystem Id: 0x104382fe
Revision Id: 0x100101
No Modem Function Group found
Default PCM:
rates [0x560]: 44100 48000 96000 192000
bits [0xe]: 16 20 24
formats [0x1]: PCM
Default Amp-In caps: N/A
Default Amp-Out caps: N/A
GPIO: io=2, o=0, i=0, unsolicited=1, wake=0
IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0
IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0
Node 0x02 [Audio Output] wcaps 0x11: Stereo
Device: name="ALC1200 Analog", type="Audio", device=0
Converter: stream=5, channel=0
PCM:
rates [0x560]: 44100 48000 96000 192000
bits [0xe]: 16 20 24
formats [0x1]: PCM
Node 0x03 [Audio Output] wcaps 0x11: Stereo
Converter: stream=5, channel=0
PCM:
rates [0x560]: 44100 48000 96000 192000
bits [0xe]: 16 20 24
formats [0x1]: PCM
Node 0x04 [Audio Output] wcaps 0x11: Stereo
Converter: stream=5, channel=0
PCM:
rates [0x560]: 44100 48000 96000 192000
bits [0xe]: 16 20 24
formats [0x1]: PCM
Node 0x05 [Audio Output] wcaps 0x11: Stereo
Converter: stream=5, channel=0
PCM:
rates [0x560]: 44100 48000 96000 192000
bits [0xe]: 16 20 24
formats [0x1]: PCM
Node 0x06 [Audio Output] wcaps 0x211: Stereo Digital
Converter: stream=5, channel=0
Digital: Enabled
Digital category: 0x0
PCM:
rates [0x5e0]: 44100 48000 88200 96000 192000
bits [0x1e]: 16 20 24 32
formats [0x1]: PCM
(...)
so, i've decompressed the .flac source into 24-bit/96kHz .wav and pushed into aplay.
$ aplay -v DvorakViolinCon.wav
Playing WAVE 'DvorakViolinCon.wav' : Signed 24 bit Little Endian in 3bytes, Rate 96000 Hz, Stereo
Plug PCM: Rate conversion PCM (48000, sformat=S32_LE)
Converter: linear-interpolation
Protocol version: 10002
Its setup is:
stream : PLAYBACK
access : RW_INTERLEAVED
format : S24_3LE
subformat : STD
channels : 2
rate : 96000
exact rate : 96000 (96000/1)
msbits : 24
buffer_size : 16384
period_size : 2048
period_time : 21333
tstamp_mode : NONE
period_step : 1
avail_min : 2048
period_event : 0
start_threshold : 16384
stop_threshold : 16384
silence_threshold: 0
silence_size : 0
boundary : 4611686018427387904
Slave: Soft volume PCM
Control: PCM Playback Volume
min_dB: -51
max_dB: 0
resolution: 256
Its setup is:
stream : PLAYBACK
access : MMAP_INTERLEAVED
format : S32_LE
subformat : STD
channels : 2
rate : 48000
exact rate : 48000 (48000/1)
msbits : 32
buffer_size : 8192
period_size : 1024
period_time : 21333
tstamp_mode : NONE
period_step : 1
avail_min : 1024
period_event : 0
start_threshold : 8192
stop_threshold : 8192
silence_threshold: 0
silence_size : 0
boundary : 4611686018427387904
Slave: Direct Stream Mixing PCM
Its setup is:
stream : PLAYBACK
access : MMAP_INTERLEAVED
format : S32_LE
subformat : STD
channels : 2
rate : 48000
exact rate : 48000 (48000/1)
msbits : 32
buffer_size : 8192
period_size : 1024
period_time : 21333
tstamp_mode : NONE
period_step : 1
avail_min : 1024
period_event : 0
start_threshold : 8192
stop_threshold : 8192
silence_threshold: 0
silence_size : 0
boundary : 4611686018427387904
Hardware PCM card 0 'HDA Intel' device 0 subdevice 0
Its setup is:
stream : PLAYBACK
access : MMAP_INTERLEAVED
format : S32_LE
subformat : STD
channels : 2
rate : 48000
exact rate : 48000 (48000/1)
msbits : 32
buffer_size : 8192
period_size : 1024
period_time : 21333
tstamp_mode : ENABLE
period_step : 1
avail_min : 1024
period_event : 0
start_threshold : 1
stop_threshold : 4611686018427387904
silence_threshold: 0
silence_size : 4611686018427387904
boundary : 4611686018427387904
appl_ptr : 0
hw_ptr : 0
hmm, why it's resampled to 48kHz? does alsa support 96/192kHz sample rates?
BR,
Pawel.
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