lossy (16-bit) af resampling on 24-bit/96kHz audio?

[pluto@…]

hi,

i've downloaded a sample flac violin concert in 24-bit/96kHz quality.
(​http://01688cb.netsolhost.com/samplerdownload/DvorakViolinCon.flac)
during playing it with 'mplayer -v' i see some weird info about resampling.

==========================================================================
Opening audio decoder: [ffmpeg] FFmpeg/libavcodec audio decoders
dec_audio: Allocating 192000 + 65536 = 257536 bytes for output buffer.
FFmpeg's libavcodec audio codec
INFO: libavcodec "flac" init OK!
[flac @ 0xc303a0] Max Blocksize: 4096
[flac @ 0xc303a0] Max Framesize: 16721
[flac @ 0xc303a0] Samplerate: 96000
[flac @ 0xc303a0] Channels: 2
[flac @ 0xc303a0] Bits: 24
AUDIO: 96000 Hz, 2 ch, s32le, 2731.6 kbit/44.46% (ratio: 341455->768000)
Selected audio codec: [ffflac] afm: ffmpeg (FFmpeg FLAC audio)
==========================================================================
Building audio filter chain for 96000Hz/2ch/s32le -> 0Hz/0ch/??...
[libaf] Adding filter dummy
[dummy] Was reinitialized: 96000Hz/2ch/s32le
[dummy] Was reinitialized: 96000Hz/2ch/s32le
Trying preferred audio driver 'alsa', options '[none]'
alsa-init: requested format: 96000 Hz, 2 channels, 19
alsa-init: using ALSA 1.0.23
alsa-init: setup for 1/2 channel(s)
alsa-init: using device default
alsa-init: pcm opened in blocking mode
alsa-init: got buffersize=65536
alsa-init: got period size 1024
alsa: 192000 Hz/2 channels/8 bpf/65536 bytes buffer/Signed 32 bit Little Endian
AO: [alsa] 192000Hz 2ch s32le (4 bytes per sample)
AO: Description: ALSA-0.9.x-1.x audio output
AO: Author: Alex Beregszaszi, Zsolt Barat <joy@…>
AO: Comment: under development
Building audio filter chain for 96000Hz/2ch/s32le -> 192000Hz/2ch/s32le...
[dummy] Was reinitialized: 96000Hz/2ch/s32le
[libaf] Adding filter lavcresample
[libaf] Adding filter format
[format] Changing sample format from little-endian 32-bit signed int to little-endian 16-bit signed int
[dummy] Was reinitialized: 192000Hz/2ch/s16le
[libaf] Adding filter format
[format] Changing sample format from little-endian 16-bit signed int to little-endian 32-bit signed int
[format] Changing sample format from little-endian 32-bit signed int to little-endian 16-bit signed int
[dummy] Was reinitialized: 192000Hz/2ch/s16le
[format] Changing sample format from little-endian 16-bit signed int to little-endian 32-bit signed int

is it really resamples from 96kHz/s32le to 192kHz/s32le with
lossy s16le in the middle? i'm observing similar log message
when playing 96kHz samples with soundcard rate defaulted to 48kHz.

my ALC1200 hda codec is defaulted to 192kHz with:

$ cat ~/.asoundrc
defaults.pcm.dmix.!rate 192000

[cehoyos]

Using an -ao alsa:device=hw=1.7 device that does support 96kHz/s32le, no resampling is done, if I use an analog audio device (and -v), I can reproduce this output.