roberth

So it then just fills up the remaining 16 bit with zeros when doing this converting?

Here you go. # An example configuration file for MPD. # Read the user manual for documentation: http://www.musicpd.org/doc/user/ # or /usr/share/doc/mpd/user-manual.html # Files and directories ####################################################### # # This setting controls the top directory which MPD will search to discover the # available audio files and add them to the daemon's online database. This # setting defaults to the XDG directory, otherwise the music directory will be # be disabled and audio files will only be accepted over ipc socket (using # file:// protocol) or streaming files over an accepted protocol. # music_directory "/mnt/DATA/sync/musikk" # # This setting sets the MPD internal playlist directory. The purpose of this # directory is storage for playlists created by MPD. The server will use # playlist files not created by the server but only if they are in the MPD # format. This setting defaults to playlist saving being disabled. # playlist_directory "/home/roberth/.config/mpd/playlists" # # This setting sets the location of the MPD database. This file is used to # load the database at server start up and store the database while the # server is not up. This setting defaults to disabled which will allow # MPD to accept files over ipc socket (using file:// protocol) or streaming # files over an accepted protocol. # db_file "/home/roberth/.config/mpd/database" # # These settings are the locations for the daemon log files for the daemon. # These logs are great for troubleshooting, depending on your log_level # settings. # # The special value "syslog" makes MPD use the local syslog daemon. This # setting defaults to logging to syslog, otherwise logging is disabled. # log_file "/home/roberth/.config/mpd/mpd.log" # # This setting sets the location of the file which stores the process ID # for use of mpd --kill and some init scripts. This setting is disabled by # default and the pid file will not be stored. # pid_file "/home/roberth/.config/mpd/pid" # # This setting sets the location of the file which contains information about # most variables to get MPD back into the same general shape it was in before # it was brought down. This setting is disabled by default and the server # state will be reset on server start up. # state_file "/home/roberth/.config/mpd/state" # # The location of the sticker database. This is a database which # manages dynamic information attached to songs. # #sticker_file "/var/lib/mpd/sticker.sql" # ############################################################################### # General music daemon options ################################################ # # This setting specifies the user that MPD will run as. MPD should never run as # root and you may use this setting to make MPD change its user ID after # initialization. This setting is disabled by default and MPD is run as the # current user. # #user "roberth" # # This setting specifies the group that MPD will run as. If not specified # primary group of user specified with "user" setting will be used (if set). # This is useful if MPD needs to be a member of group such as "audio" to # have permission to use sound card. # #group "users" # # This setting sets the address for the daemon to listen on. Careful attention # should be paid if this is assigned to anything other then the default, any. # This setting can deny access to control of the daemon. Choose any if you want # to have mpd listen on every address # # For network bind_to_address "localhost" # # And for Unix Socket #bind_to_address "/run/mpd/socket" # # This setting is the TCP port that is desired for the daemon to get assigned # to. # port "6600" # # This setting controls the type of information which is logged. Available # setting arguments are "default", "secure" or "verbose". The "verbose" setting # argument is recommended for troubleshooting, though can quickly stretch # available resources on limited hardware storage. # log_level "verbose" # # If you have a problem with your MP3s ending abruptly it is recommended that # you set this argument to "no" to attempt to fix the problem. If this solves # the problem, it is highly recommended to fix the MP3 files with vbrfix # (available as vbrfix in the debian archive), at which # point gapless MP3 playback can be enabled. # #gapless_mp3_playback "yes" # # Setting "restore_paused" to "yes" puts MPD into pause mode instead # of starting playback after startup. # #restore_paused "no" # # This setting enables MPD to create playlists in a format usable by other # music players. # #save_absolute_paths_in_playlists "no" # # This setting defines a list of tag types that will be extracted during the # audio file discovery process. The complete list of possible values can be # found in the mpd.conf man page. #metadata_to_use "artist,album,title,track,name,genre,date,composer,performer,disc" # # This setting enables automatic update of MPD's database when files in # music_directory are changed. # auto_update "yes" # # Limit the depth of the directories being watched, 0 means only watch # the music directory itself. There is no limit by default. # #auto_update_depth "3" # ############################################################################### # Symbolic link behavior ###################################################### # # If this setting is set to "yes", MPD will discover audio files by following # symbolic links outside of the configured music_directory. # #follow_outside_symlinks "yes" # # If this setting is set to "yes", MPD will discover audio files by following # symbolic links inside of the configured music_directory. # #follow_inside_symlinks "yes" # ############################################################################### # Zeroconf / Avahi Service Discovery ########################################## # # If this setting is set to "yes", service information will be published with # Zeroconf / Avahi. # #zeroconf_enabled "yes" # # The argument to this setting will be the Zeroconf / Avahi unique name for # this MPD server on the network. # #zeroconf_name "Music Player" # ############################################################################### # Permissions ################################################################# # # If this setting is set, MPD will require password authorization. The password # can setting can be specified multiple times for different password profiles. # #password "password@read,add,control,admin" # # This setting specifies the permissions a user has who has not yet logged in. # #default_permissions "read,add,control,admin" # ############################################################################### # Database ####################################################################### # #database { # plugin "proxy" # host "other.mpd.host" # port "6600" #} # Input ####################################################################### # input { plugin "curl" # proxy "proxy.isp.com:8080" # proxy_user "user" # proxy_password "password" } # ############################################################################### # Audio Output ################################################################ # # MPD supports various audio output types, as well as playing through multiple # audio outputs at the same time, through multiple audio_output settings # blocks. Setting this block is optional, though the server will only attempt # autodetection for one sound card. # # An example of an ALSA output: # audio_output { type "alsa" name "My ALSA Device" device "hw:2,0" auto_resample "no" mixer_type "no" auto_format "no" } # # An example of an OSS output: # #audio_output { # type "oss" # name "My OSS Device" # device "/dev/dsp" # optional # mixer_type "hardware" # optional # mixer_device "/dev/mixer" # optional # mixer_control "PCM" # optional #} # # An example of a shout output (for streaming to Icecast): # #audio_output { # type "shout" # encoding "ogg" # optional # name "My Shout Stream" # host "localhost" # port "8000" # mount "/mpd.ogg" # password "hackme" # quality "5.0" # bitrate "128" # format "44100:16:1" # protocol "icecast2" # optional # user "source" # optional # description "My Stream Description" # optional # url "http://example.com" # optional # genre "jazz" # optional # public "no" # optional # timeout "2" # optional # mixer_type "software" # optional #} # # An example of a recorder output: # #audio_output { # type "recorder" # name "My recorder" # encoder "vorbis" # optional, vorbis or lame # path "/var/lib/mpd/recorder/mpd.ogg" ## quality "5.0" # do not define if bitrate is defined # bitrate "128" # do not define if quality is defined # format "44100:16:1" #} # # An example of a httpd output (built-in HTTP streaming server): # #audio_output { # type "httpd" # name "My HTTP Stream" # encoder "vorbis" # optional, vorbis or lame # port "8000" # bind_to_address "0.0.0.0" # optional, IPv4 or IPv6 # quality "5.0" # do not define if bitrate is defined # bitrate "128" # do not define if quality is defined # format "44100:16:1" # max_clients "0" # optional 0=no limit #} # # An example of a pulseaudio output (streaming to a remote pulseaudio server) # Please see README.Debian if you want mpd to play through the pulseaudio # daemon started as part of your graphical desktop session! # #audio_output { # type "pulse" # name "My Pulse Output" # server "remote_server" # optional # sink "remote_server_sink" # optional #} # # An example of a winmm output (Windows multimedia API). # #audio_output { # type "winmm" # name "My WinMM output" # device "Digital Audio (S/PDIF) (High Definition Audio Device)" # optional # or # device "0" # optional # mixer_type "hardware" # optional #} # # An example of an openal output. # #audio_output { # type "openal" # name "My OpenAL output" # device "Digital Audio (S/PDIF) (High Definition Audio Device)" # optional #} # ## Example "pipe" output: # #audio_output { # type "pipe" # name "my pipe" # command "aplay -f cd 2>/dev/null" ## Or if you're want to use AudioCompress # command "AudioCompress -m | aplay -f cd 2>/dev/null" ## Or to send raw PCM stream through PCM: # command "nc example.org 8765" # format "44100:16:2" #} # ## An example of a null output (for no audio output): # #audio_output { # type "null" # name "My Null Output" # mixer_type "none" # optional #} # # If MPD has been compiled with libsamplerate support, this setting specifies # the sample rate converter to use. Possible values can be found in the # mpd.conf man page or the libsamplerate documentation. By default, this is # setting is disabled. # #samplerate_converter "Fastest Sinc Interpolator" # ############################################################################### # Normalization automatic volume adjustments ################################## # # This setting specifies the type of ReplayGain to use. This setting can have # the argument "off", "album", "track" or "auto". "auto" is a special mode that # chooses between "track" and "album" depending on the current state of # random playback. If random playback is enabled then "track" mode is used. # See <http://www.replaygain.org> for more details about ReplayGain. # This setting is off by default. # #replaygain "album" # # This setting sets the pre-amp used for files that have ReplayGain tags. By # default this setting is disabled. # #replaygain_preamp "0" # # This setting sets the pre-amp used for files that do NOT have ReplayGain tags. # By default this setting is disabled. # #replaygain_missing_preamp "0" # # This setting enables or disables ReplayGain limiting. # MPD calculates actual amplification based on the ReplayGain tags # and replaygain_preamp / replaygain_missing_preamp setting. # If replaygain_limit is enabled MPD will never amplify audio signal # above its original level. If replaygain_limit is disabled such amplification # might occur. By default this setting is enabled. # #replaygain_limit "yes" # # This setting enables on-the-fly normalization volume adjustment. This will # result in the volume of all playing audio to be adjusted so the output has # equal "loudness". This setting is disabled by default. # #volume_normalization "no" # ############################################################################### # Character Encoding ########################################################## # # If file or directory names do not display correctly for your locale then you # may need to modify this setting. # filesystem_charset "UTF-8" # # This setting controls the encoding that ID3v1 tags should be converted from. # id3v1_encoding "UTF-8" # ############################################################################### # SIDPlay decoder ############################################################# # # songlength_database: # Location of your songlengths file, as distributed with the HVSC. # The sidplay plugin checks this for matching MD5 fingerprints. # See http://www.c64.org/HVSC/DOCUMENTS/Songlengths.faq # # default_songlength: # This is the default playing time in seconds for songs not in the # songlength database, or in case you're not using a database. # A value of 0 means play indefinitely. # # filter: # Turns the SID filter emulation on or off. # #decoder { # plugin "sidplay" # songlength_database "/media/C64Music/DOCUMENTS/Songlengths.txt" # default_songlength "120" # filter "true" #} # ############################################################################### audio_buffer_size "1024" buffer_before_play "10%"

Uhm... I am playing 16bit 44,1khz audio, the dac supports up to 32 bit.

Hello I have a Dac which uses the XMOS usb audio 2.0 chip, and I use linux and MPD to play music with it. So I am trying to achieve bitperfect playback. I am not sure if I am achieving that. When I play a song, MPD shows this in the log: [font=monospace][color=#000000]Apr 21 07:46 : playlist: play 0:"10cc/Bloody Tourists/01 - 10CC - Dreadlock Holiday.flac"[/color] Apr 21 07:46 : client: [0] command returned 0 Apr 21 07:46 : decoder_thread: probing plugin flac Apr 21 07:46 : playlist: queue song 1:"Twisted Sister/Stay Hungry/05 - I Wanna Rock.flac" Apr 21 07:46 : decoder: audio_format=44100:16:2, seekable=true Apr 21 07:46 : client: [0] process command "status" Apr 21 07:46 : alsa_output: opened hw:2,0 type=HW Apr 21 07:46 : alsa_output: format=S32_LE (Signed 32 bit Little Endian) Apr 21 07:46 : alsa_output: buffer: size=16..131072 time=362..2972155 Apr 21 07:46 : alsa_output: period: size=8..65536 time=181..1486078 Apr 21 07:46 : alsa_output: default period_time = buffer_time/4 = 500000/4 = 125000 Apr 21 07:46 : alsa_output: buffer_size=22050 period_size=5513 Apr 21 07:46 : output: opened plugin=alsa name="My ALSA Device" audio_format=44100:32:2 Apr 21 07:46 : output: converting from 44100:16:2 [/font] Does this info indicate that it resample the audio? Some info: [font=monospace][color=#000000]roberth@Magda [/color][color=#18B218]~[/color][color=#000000]> [/color][color=#005FD7]aplay[/color][color=#00AFFF]-l[/color] **** List of PLAYBACK Hardware Devices **** card 0: HDMI [HDA Intel HDMI], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: HDMI [HDA Intel HDMI], device 7: HDMI 1 [HDMI 1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: HDMI [HDA Intel HDMI], device 8: HDMI 2 [HDMI 2] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: PCH [HDA Intel PCH], device 0: ALC1150 Analog [ALC1150 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: PCH [HDA Intel PCH], device 1: ALC1150 Digital [ALC1150 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: X20 [XMOS USB Audio 2.0], device 0: USB Audio [uSB Audio] Subdevices: 0/1 Subdevice #0: subdevice #0 [/font] [font=monospace][color=#000000]roberth@Magda [/color][color=#18B218]~[/color][color=#000000]> [/color][color=#005FD7]aplay[/color][color=#00AFFF]-L[/color] null Discard all samples (playback) or generate zero samples (capture) hdmi:CARD=HDMI,DEV=0 HDA Intel HDMI, HDMI 0 HDMI Audio Output hdmi:CARD=HDMI,DEV=1 HDA Intel HDMI, HDMI 1 HDMI Audio Output hdmi:CARD=HDMI,DEV=2 HDA Intel HDMI, HDMI 2 HDMI Audio Output sysdefault:CARD=PCH HDA Intel PCH, ALC1150 Analog Default Audio Device front:CARD=PCH,DEV=0 HDA Intel PCH, ALC1150 Analog Front speakers surround21:CARD=PCH,DEV=0 HDA Intel PCH, ALC1150 Analog 2.1 Surround output to Front and Subwoofer speakers surround40:CARD=PCH,DEV=0 HDA Intel PCH, ALC1150 Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=PCH,DEV=0 HDA Intel PCH, ALC1150 Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=PCH,DEV=0 HDA Intel PCH, ALC1150 Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=PCH,DEV=0 HDA Intel PCH, ALC1150 Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=PCH,DEV=0 HDA Intel PCH, ALC1150 Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers iec958:CARD=PCH,DEV=0 HDA Intel PCH, ALC1150 Digital IEC958 (S/PDIF) Digital Audio Output sysdefault:CARD=X20 XMOS USB Audio 2.0, USB Audio Default Audio Device front:CARD=X20,DEV=0 XMOS USB Audio 2.0, USB Audio Front speakers surround21:CARD=X20,DEV=0 XMOS USB Audio 2.0, USB Audio 2.1 Surround output to Front and Subwoofer speakers surround40:CARD=X20,DEV=0 XMOS USB Audio 2.0, USB Audio 4.0 Surround output to Front and Rear speakers surround41:CARD=X20,DEV=0 XMOS USB Audio 2.0, USB Audio 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=X20,DEV=0 XMOS USB Audio 2.0, USB Audio 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=X20,DEV=0 XMOS USB Audio 2.0, USB Audio 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=X20,DEV=0 XMOS USB Audio 2.0, USB Audio 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers iec958:CARD=X20,DEV=0 XMOS USB Audio 2.0, USB Audio IEC958 (S/PDIF) Digital Audio Output [/font] MPD output configuration: [font=monospace][color=#000000]audio_output {[/color] type "alsa" name "My ALSA Device" device "hw:2,0" auto_resample "no" mixer_type "no" auto_format "no" [/font] I also noticed through alsamixer that it uses a mixer that controls the volume when using mpd, is that neccesary to disable?