.wav format doesn't show (kbps number)

Forums:

Hello MOC staff/devs!!!

For some time i have noticed "WAV Files" doesn't show kbp/s numbers below window.
Only on WAV format have see something like this missing.

FLAC, APE, MP3... are showing normally.

here it is
IMAGE(http://i.imgur.com/mhejGx1.png)

I do not mind posting, simply a cosmetic issue but can you check a ".wav" files please?

$ nili@debian:~$ apt-cache policy moc

moc:
Installed: 1:2.5.0-1
Candidate: 1:2.5.0-1
Version table:
*** 1:2.5.0-1 0
500 http://ftp.us.debian.org/debian/ jessie/main i386 Packages
100 /var/lib/dpkg/status

Regards,
Nili

Wav is the industry standard for uncompressed lossless audio. Yes 1,411kbps is the raw bit rate for a audio CD. All audio CDs that you will burn will come out as 1411kbps music files. A Wav file will fluctuate in size depending on the source material you are encoding. For example, (without getting into to much detail) an audio CD like what you buy in the stores, all have a maximum resolution of 16-bit 44,100HZ. Doing the math, when that is encoded into Wav it will give you a bit rate of 1,411kbps. But if I we're to rip audio of off of a DVD which has a resolution of 16-bit at a slightly higher sample rate of 48,000HZ I would get a Wav file that is abut 1,560kbps (Give or take) So It just depends on the native audio resolution of your source material.

Elimar

Hi riesebie,
Thanks for explanation and clarification of some things, in fact, a part of your message is known to me. Those wav from the image ^above are made rip from SACD DVD

Originally they were up to 5000 kbps, I converted to WAV with foobar2000 , after converting they dropped to 2822kbp/s with output 32-bit selected. So, now i have them on 2822kbp/s as .wav

I know some PCM WAV shown as 1436kbp/s usually from DVD's some other like you said 1411kbps. Apparently WAV shown different to the manner how it is done the converting, 1411 is standart from what i know.

Wish MOC to read all possibility of WAV kbps even if is static/standard or converted by user.

I remember some other audio player can show those wav kbps... MOC is my main favorite audio player, I would use even if stay empty that part of info.

It should not be difficult to correct it for PCM files at least with sndfile decoder. I'll try to prepare a patch. However, what you probably rather want is a possibility to see details about sound format (sampling rate, bit depth, ...)

Let me also add that the term "SACD DVD" is very confusing, as SACDs are not DVD disks, unlike DVD-A. Did something change and SACDs became rippable, or are you refering just to an re-recording of a SACD via analog connection? In any case, storing the result in 32-bit file offers no benefit over 16-bits, see e.g. http://people.xiph.org/~xiphmont/demo/neil-young.html

That's only half of the story as .wav allows also compressed audio (even mp3), thus doing the math won't always help. See for example wikipedia: https://en.wikipedia.org/wiki/WAV#WAV_file_audio_coding_formats_compared

Thanks for clarification.

The MOC decoder plug-ins use the values they get from the applicable library to calculate the kbps value (amongst others). But unfortunately in some cases the library returns garbage in those values. MOC detects this as best it's able and, rather than perpetuating GIGO, simply blanks what would be an otherwise meaningless value.

For the FFmpeg/LibAV library, breakages occur for the WAV format (amongst others) so I suspect this is the cause of your problem. MOC's doing its best, but, short of just guessing, it can't do more than that.

(Those interested in the exact details can look at is_timing_broken() in the FFmpeg decoder.)

As a circumvention, you could try the sndfile decoder (if you have it installed) for WAV files via the MOC PreferredDecoders configuration option (if you've changed it from the default).

(Some of that assumes you are using the FFmpeg decoder for WAV files, of course.)

Hi jcf,
I use more or less in default settings with very few changes, here's the config i'm using:

# This is a configuration file for the MOC player. It should be named<br /> # 'config' and placed in the ~/.moc directory. As this file can specify<br /> # commands which invoke other applications, MOC will refuse to start if it<br /> # is not owned by either root or the current user, or if it is writable by<br /> # anyone other than its owner. All options are given with their default<br /> # values, and therefore commented.</p> <p># Comments begin with '#'.<br /> # You can use quotes and escape ('\') in parameters.<br /> #<br /> # You can have variable values substituted by enclosing the variable name<br /> # as "${...}". (This only applies to the portion of the option following<br /> # the ' ='.) Variables are substituted first from the environment then,<br /> # if not found, from the configuration options. (Note that the value of<br /> # a configuration option substituted is that which it has at the time the<br /> # substitution variable is encountered.) If there is a naming conflict<br /> # between an environment and configuration variable, you may be able to<br /> # resolve it by using lowercase as the environment variable matches are<br /> # case-sensitive whereas the configuration variables are not.<br /> #<br /> # You can also use the form "${...:-...}" where the value in the second<br /> # position will be substituted if the variable name given in the first<br /> # position is unset or null.<br /> #<br /> # So, for example:<br /> #<br /> # MusicDir = /music/${USER:-public}<br /> # Fastdir1 = ${MusicDir}/mp3/rock<br /> # Fastdir2 = ${MusicDir}/mp3/electronic<br /> # Fastdir3 = ${MusicDir}/mp3/rap<br /> # Fastdir4 = ${MusicDir}/mp3/etc<br /> #<br /> # Variable names are limited to those accepted by the BASH shell; that<br /> # is, those comprising the upper- and lowercase ASCII characters, digits<br /> # and the underscore.<br /> #<br /> # If you need to use the "${" sequence for any other purpose, write "$${"<br /> # and it will be replaced by "${" and not treated as a substitution.<br /> #<br /> # Some options take lists of strings as their values. The strings are<br /> # separated by colons. Additional strings can be appended to the list<br /> # using "+ =" in place of a plain "=" to assign the value. For an example,<br /> # see the XTerms option.<br /> #<br /> # You can override any configuration option when you run MOC using the<br /> # '-O' command line option:<br /> #<br /> # mocp -O AutoNext =no -O messagelingertime=1 -O XTerms+=xxt:xwt<br /> #<br /> # This command line option can be repeated as many times as needed and<br /> # the configuration option name is not case sensitive. (Note that MOC<br /> # does not perform variable substitution on the value of such overridden<br /> # configuration options.) Most option values are set before the<br /> # configuration file is processed (which allows the new values to be<br /> # picked up by substitutions), however list-valued options are overridden<br /> # afterwards (which gives the choice of whether the configured values are<br /> # replaced or added to).</p> <p># Remember that the client and server are separate processes and the<br /> # server will retain the configuration values formed from the environment<br /> # within which it was originally started.</p> <p># Show file titles (title, author, album) instead of file names?<br /> ReadTags = no</p> <p># In which directory do you store your music files? If you specify it<br /> # you will be able to jump straight to this directory with the '-m'<br /> # parameter or the 'm' command. This can also point to a playlist.<br /> #<br /> # Example: MusicDir = "/home/joe/music"<br /> #<br /> #MusicDir =</p> <p># Start in the music directory by default? If set to 'no', start<br /> # in the current directory by default. A single directory on<br /> # the command line takes precedence.<br /> #StartInMusicDir = no</p> <p># How to sort? FileName is the option's only value for now.<br /> #Sort = FileName</p> <p># Show errors in the streams (for example, broken frames in MP3 files)?<br /> #ShowStreamErrors = no</p> <p># Ignore CRC errors in MP3 files? Most players do that, so the default<br /> # value is 'yes'.<br /> #MP3IgnoreCRCErrors = yes</p> <p># Set playback toggles.<br /> #Repeat = no<br /> #Shuffle = no<br /> #AutoNext = yes</p> <p># Default FormatString:<br /> #<br /> # %n - Track number<br /> # %a - Artist<br /> # %A - Album<br /> # %t - Title<br /> # %(X:TRUE:FALSE) - Ternary expression: if X exists, do TRUE,<br /> # otherwise FALSE. The escape character must<br /> # be doubled (i.e., '\\'). (See zshmisc<br /> # documentation for more information.)<br /> #<br /> #FormatString = "%(n:%n :)%(a:%a - :)%(t:%t:)%(A: \(%A\):)"</p> <p># Input and output buffer sizes (in kilobytes).<br /> #InputBuffer = 512 # Minimum value is 32KB<br /> #OutputBuffer = 512 # Minimum value is 128KB</p> <p># How much to fill the input buffer before playing (in kilobytes)?<br /> # This can't be greater than the value of InputBuffer. While this has<br /> # a positive effect for network streams, it also causes the broadcast<br /> # audio to be delayed.<br /> #Prebuffering = 64</p> <p># Use this HTTP proxy server for internet streams. If not set, the<br /> # environment variables http_proxy and ALL_PROXY will be used if present.<br /> #<br /> # Format: HTTPProxy = PROXY_NAME:PORT<br /> #<br /> #HTTPProxy =</p> <p># Sound driver - OSS, ALSA, JACK, SNDIO (on OpenBSD) or null (only for<br /> # debugging). You can enter more than one driver as a colon-separated<br /> # list. The first working driver will be used.<br /> #SoundDriver = JACK:ALSA:OSS</p> <p># Jack output settings.<br /> #JackClientName = "moc"<br /> #JackStartServer = no<br /> #JackOutLeft = "system:playback_1"<br /> #JackOutRight = "system:playback_2"</p> <p># OSS output settings.<br /> #OSSDevice = /dev/dsp<br /> #OSSMixerDevice = /dev/mixer<br /> #OSSMixerChannel1 = pcm # 'pcm', 'master' or 'speaker'<br /> #OSSMixerChannel2 = master # 'pcm', 'master' or 'speaker'</p> <p># ALSA output settings. If you need to dump the audio produced by MOC<br /> # to a file for diagnostic purposes, the following setting of 'ALSADevice'<br /> # should do that:<br /> #<br /> # ALSADevice=tee:hw,'/tmp/out.wav',wav<br /> #<br /> #ALSADevice = default<br /> #ALSAMixer1 = PCM<br /> #ALSAMixer2 = Master</p> <p># Save software mixer state?<br /> # If enabled, a file 'softmixer' will be created in '~/.moc/' storing the<br /> # mixersetting set when the server is shut down.<br /> # Note that there is a "hidden" 'Amplification' setting in that file.<br /> # Amplification (0-200) is used to scale the mixer setting (0-100). This<br /> # results in a higher signal amplitude but may also produce clipping.<br /> #Softmixer_SaveState = yes</p> <p># Save equalizer state?<br /> # If enabled, a file 'equalizer' will be created in '~/.moc/' storing the<br /> # equalizer settings when the server is shut down.<br /> # Note that there is a "hidden" 'Mixin' setting in that file.<br /> # Mixin (0.0-1.0) is used to determine how much of the original signal is<br /> # used after equalizing. 0 means to only use the equalized sound, while 1<br /> # effectively disabled the mixer. The default is 0.25.<br /> #Equalizer_SaveState = yes</p> <p># Show files with dot at the beginning?<br /> #ShowHiddenFiles = no</p> <p># Hide file name extensions?<br /> #HideFileExtension = no</p> <p># Show file format in menu?<br /> #ShowFormat = yes</p> <p># Show file time in menu? Possible values: 'yes', 'no' and 'IfAvailable'<br /> # (meaning show the time only when it is already known, which often works<br /> # faster).<br /> #ShowTime = IfAvailable</p> <p># Show time played as a percentage in the time progress bar.<br /> #ShowTimePercent = no</p> <p># Values of the TERM environment variable which are deemed to be xterms.<br /> #XTerms = xterm<br /> #XTerms + = xterm-colour:xterm-color<br /> #XTerms + = xterm-256colour:xterm-256color<br /> #XTerms + = rxvt:rxvt-unicode<br /> #XTerms + = rxvt-unicode-256colour:rxvt-unicode-256color<br /> #XTerms + = eterm</p> <p># Theme file to use. This can be absolute path or relative to<br /> # /usr/share/moc/themes/ (depends on installation prefix) or<br /> # ~/.moc/themes/ .<br /> #<br /> # Example: Theme = laras_theme<br /> #<br /> Theme = blue_theme</p> <p># The theme used when running on an xterm.<br /> #<br /> # Example: XTermTheme = transparent-background<br /> #<br /> #XTermTheme =</p> <p># Should MOC try to autoload the default lyrics file for an audio? (The<br /> # default lyrics file is a text file with the same file name as the audio<br /> # file name with any trailing "extension" removed.)<br /> #AutoLoadLyrics = yes</p> <p># MOC directory (where pid file, socket and state files are stored).<br /> # You can use ~ at the beginning.<br /> #MOCDir = ~/.moc</p> <p># Use mmap() to read files. mmap() is much slower on NFS.<br /> #UseMMap = no</p> <p># Use MIME to identify audio files. This can make for slower loading<br /> # of playlists but is more accurate than using "extensions".<br /> #UseMimeMagic = no</p> <p># Assume this encoding for ID3 version 1/1.1 tags (MP3 files). Unlike<br /> # ID3v2, UTF-8 is not used here and MOC can't guess how tags are encoded.<br /> # Another solution is using librcc (see the next option). This option is<br /> # ignored if UseRCC is set to 'yes'.<br /> #ID3v1TagsEncoding = WINDOWS-1250</p> <p># Use librcc to fix ID3 version 1/1.1 tags encoding.<br /> #UseRCC = yes</p> <p># Use librcc to filenames and directory names encoding.<br /> #UseRCCForFilesystem = yes</p> <p># When this option is set the player assumes that if the encoding of<br /> # ID3v2 is set to ISO-8859-1 then the ID3v1TagsEncoding is actually<br /> # that and applies appropriate conversion.<br /> #EnforceTagsEncoding = no</p> <p># Enable the conversion of filenames from the local encoding to UTF-8.<br /> #FileNamesIconv = no</p> <p># Enable the conversion of the xterm title from UTF-8 to the local encoding.<br /> #NonUTFXterm = no</p> <p># Remember the playlist after exit?<br /> #SavePlaylist = yes</p> <p># When using more than one client (interface) at a time, do they share<br /> # the playlist?<br /> #SyncPlaylist = yes</p> <p># Choose a keymap file (relative to '~/.moc/' or using an absolute path).<br /> # An annotated example keymap file is included ('keymap.example').<br /> #<br /> # Example: Keymap = my_keymap<br /> #<br /> #Keymap =</p> <p># Use ASCII rather than graphic characters for drawing lines. This<br /> # helps on some terminals.<br /> #ASCIILines = no</p> <p># FastDirs, these allow you to jump directly to a directory, the key<br /> # bindings are in the keymap file.<br /> #<br /> # Examples: Fastdir1 = /mp3/rock<br /> # Fastdir2 = /mp3/electronic<br /> # Fastdir3 = /mp3/rap<br /> # Fastdir4 = /mp3/etc<br /> #<br /> #Fastdir1 =<br /> #Fastdir2 =<br /> #Fastdir3 =<br /> #Fastdir4 =<br /> #Fastdir5 =<br /> #Fastdir6 =<br /> #Fastdir7 =<br /> #Fastdir8 =<br /> #Fastdir9 =<br /> #Fastdir10 =</p> <p># How fast to seek (in number of seconds per keystroke). The first<br /> # option is for normal seek and the second for silent seek.<br /> #SeekTime = 1<br /> #SilentSeekTime = 5</p> <p># PreferredDecoders allows you to specify which decoder should be used<br /> # for any given audio format. It is a colon-separated list in which<br /> # each entry is of the general form 'code(decoders)', where 'code'<br /> # identifies the audio format and 'decoders' is a comma-separated list<br /> # of decoders in order of preference.<br /> #<br /> # The audio format identifier may be either a filename extension or a<br /> # MIME media type. If the latter, the format is 'type/subtype' (e.g.,<br /> # 'audio/flac'). Because different systems may give different MIME<br /> # media types, any 'x-' prefix of the subtype is ignored both here and<br /> # in the actual file MIME type (so all combinations of 'audio/flac' and<br /> # 'audio/x-flac' match each other).<br /> #<br /> # For Internet streams the matching is done on MIME media type and on<br /> # actual content. For files the matches are made on MIME media type<br /> # (if the 'UseMimeMagic' option is set) and on filename extension. The<br /> # MIME media type of a file is not determined until the first entry for<br /> # MIME is encountered in the list.<br /> #<br /> # The matching is done in the order of appearance in the list with any<br /> # entries added from the command line being matched before those listed<br /> # here. Therefore, if you place all filename extension entries before<br /> # all MIME entries you will speed up MOC's processing of directories<br /> # (which could be significant for remote file systems).<br /> #<br /> # The decoder list may be empty, in which case no decoders will be used<br /> # for files (and files with that audio format ignored) while Internet<br /> # streams will be assessed on the actual content. Any decoder position<br /> # may contain an asterisk, in which case any decoder not otherwise listed<br /> # which can handle the audio format will be used. It is not an error to<br /> # list the same decoder twice, but neither does it make sense to do so.<br /> #<br /> # If you have a mix of audio and non-audio files in your directories, you<br /> # may wish to include entries at top of the list which ignore non-audio<br /> # files by extension.<br /> #<br /> # In summary, the PreferredDecoders option provides fine control over the<br /> # type of matching which is performed (filename extension, MIME media<br /> # type and streamed media content) and which decoder(s) (if any) are used<br /> # based on the option's list entries and their ordering.<br /> #<br /> # Examples: aac(aac,ffmpeg) first try FAAD2 for AACs then FFmpeg<br /> # mp3() ignore MP3 files<br /> # wav(*,sndfile) use sndfile for WAV as a last resort<br /> # ogg(vorbis,*):flac(flac,*) try Xiph decoders first<br /> # ogg():audio/ogg() ignore OGG files, and<br /> # force Internet selection by content<br /> # gz():html() ignore some non-audio files<br /> #<br /> # Any unspecified audio formats default to trying all decoders.<br /> # Any unknown (or misspelt) drivers are ignored.<br /> # All names are case insensitive.<br /> # The default setting reflects the historical situation modified by<br /> # the experience of users.<br /> #<br /> #PreferredDecoders = aac(aac,ffmpeg):m4a(ffmpeg)<br /> #PreferredDecoders += mpc(musepack,*,ffmpeg):mpc8(musepack,*,ffmpeg)<br /> #PreferredDecoders += sid(sidplay2):mus(sidplay2)<br /> #PreferredDecoders += wav(sndfile,*,ffmpeg)<br /> #PreferredDecoders += wv(wavpack,*,ffmpeg)<br /> #PreferredDecoders += audio/aac(aac):audio/aacp(aac):audio/m4a(ffmpeg)<br /> #PreferredDecoders += audio/wav(sndfile,*)</p> <p># The following PreferredDecoders attempt to handle the ambiguity surrounding<br /> # container types such as OGG for files. The first two entries will force<br /> # a local file to the correct decoder (assuming the .ogg file contains Vorbis<br /> # audio), while the MIME media types will cause Internet audio streams to<br /> # be assessed on content (which may be either Vorbis or Speex).<br /> #<br /> #PreferredDecoders += ogg(vorbis,ffmpeg):oga(vorbis,ffmpeg):ogv(ffmpeg)<br /> #PreferredDecoders += opus(opus,ffmpeg)<br /> #PreferredDecoders += spx(speex)<br /> #PreferredDecoders += application/ogg(vorbis):audio/ogg(vorbis)</p> <p># Which resampling method to use. There are a few methods of resampling<br /> # sound supported by libresamplerate. The default is 'Linear') which is<br /> # also the fastest. A better description can be found at:<br /> #<br /> # http://www.mega-nerd.com/libsamplerate/api_misc.html#Converters<br /> #<br /> # but briefly, the following methods are based on bandlimited interpolation<br /> # and are higher quality, but also slower:<br /> #<br /> # SincBestQuality - really slow (I know you probably have an xx GHz<br /> # processor, but it's still not enough to not see<br /> # this in the top output :) The worst case<br /> # Signal-to-Noise Ratio is 97dB.<br /> # SincMediumQuality - much faster.<br /> # SincFastest - the fastest bandlimited interpolation.<br /> #<br /> # And these are lower quality, but much faster methods:<br /> #<br /> # ZeroOrderHold - really poor quality, but it's really fast.<br /> # Linear - a bit better and a bit slower.<br /> #<br /> #ResampleMethod = Linear</p> <p># Always use this sample rate (in Hz) when opening the audio device (and<br /> # resample the sound if necessary). When set to 0 the device is opened<br /> # with the file's rate.<br /> #ForceSampleRate = 0</p> <p># By default, even if the sound card reports that it can output 24bit samples<br /> # MOC converts 24bit PCM to 16bit. Setting this option to 'yes' allows MOC<br /> # to use 24bit output. (The MP3 decoder, for example, uses this format.)<br /> # This is disabled by default because there were reports that it prevents<br /> # MP3 files from playing on some soundcards.<br /> #Allow24bitOutput = no</p> <p># Use realtime priority for output buffer thread. This will prevent gaps<br /> # while playing even with heavy load. The user who runs MOC must have<br /> # permissions to set such a priority. This could be dangerous, because it<br /> # is possible that a bug in MOC will freeze your computer.<br /> #UseRealtimePriority = no</p> <p># The number of audio files for which MOC will cache tags. When this limit<br /> # is reached, file tags are discarded on a least recently used basis (with<br /> # one second resolution). You can disable the cache by giving it a size of<br /> # zero. Note that if you decrease the cache size below the number of items<br /> # currently in the cache, the number will not decrease immediately (if at<br /> # all).<br /> #TagsCacheSize = 256</p> <p># Number items in the playlist.<br /> PlaylistNumbering = no</p> <p># Main window layouts can be configured. You can change the position and<br /> # size of the menus (directory and playlist). You have three layouts and<br /> # can switch between then using the 'l' key (standard mapping). By default,<br /> # only two layouts are configured.<br /> #<br /> # The format is as follows:<br /> #<br /> # - Each layout is described as a list of menu entries.<br /> # - Each menu entry is of the form:<br /> #<br /> # menu(position_x, position_y, width, height)<br /> #<br /> # where 'menu' is either 'directory' or 'playlist'.<br /> # - The parameters define position and size of the menu. They can<br /> # be absolute numbers (like 10) or a percentage of the screen size<br /> # (like 45%).<br /> # - 'width' and 'height' can have also value of 'FILL' which means<br /> # fill the screen from the menu's position to the border.<br /> # - Menus may overlap.<br /> #<br /> # You must describe at least one menu (default is to fill the whole window).<br /> # There must be at least one layout (Layout1) defined; others can be empty.<br /> #<br /> # Example: Layout1 = playlist(50%,50%,50%,50%)<br /> # Layout2 = ""<br /> # Layout3 = ""<br /> #<br /> # Just one layout, the directory will occupy the whole<br /> # screen, the playlist will have 1/4 of the screen size<br /> # and be positioned at lower right corner. (Note that<br /> # because the playlist will be hidden by the directory<br /> # you will have to use the TAB key to make the playlist<br /> # visible.)<br /> #<br /> # Example: Layout1 = playlist(0,0,100%,10):directory(0,10,100%,FILL)<br /> #<br /> # The screen is split into two parts: playlist at the top<br /> # and the directory menu at the bottom. Playlist will<br /> # occupy 10 lines and the directory menu the rest.<br /> #<br /> #Layout1 = directory(0,0,50%,100%):playlist(50%,0,FILL,100%)<br /> #Layout2 = directory(0,0,100%,100%):playlist(0,0,100%,100%)<br /> #Layout3 = ""</p> <p># When the song changes, should the menu be scrolled so that the currently<br /> # played file is visible?<br /> #FollowPlayedFile = yes</p> <p># What to do if the interface was started and the server is already playing<br /> # something from the playlist? If CanStartInPlaylist is set to 'yes', the<br /> # interface will switch to the playlist. When set to 'no' it will start<br /> # from the last directory.<br /> #CanStartInPlaylist = yes</p> <p># Executing external commands (1 - 10) invoked with key commands (F1 - F10<br /> # by default).<br /> #<br /> # Some arguments are substituted before executing:<br /> #<br /> # %f - file path<br /> # %i - title made from tags<br /> # %S - start block mark (in seconds)<br /> # %E - end block mark (in seconds)<br /> #<br /> # Data from tags can also be substituted:<br /> #<br /> # %t - title<br /> # %a - album<br /> # %r - artist<br /> # %n - track<br /> # %m - time of the file (in seconds)<br /> #<br /> # The parameters above apply to the currently selected file. If you change<br /> # them to capital letters, they are taken from the file currently playing.<br /> #<br /> # Programs are run using execv(), not a shell, so you can't do things like<br /> # redirecting the output to a file. The command string is split using blank<br /> # characters as separators; the first element is the command to be executed<br /> # and the rest are its parameters, so if you use "echo Playing: %I" we run<br /> # program 'echo' (from $PATH) with 2 parameters: the string 'Playing:' and<br /> # the title of the file currently playing. Even if the title contains<br /> # spaces, it's still one parameter and it's safe if it contains `rm -rf /`.<br /> #<br /> # Examples: ExecCommand1 = "cp %f /mnt/usb_drive"<br /> # ExecCommand2 = "/home/joe/now_playing %I"<br /> #<br /> #ExecCommand1 =<br /> #ExecCommand2 =<br /> #ExecCommand3 =<br /> #ExecCommand4 =<br /> #ExecCommand5 =<br /> #ExecCommand6 =<br /> #ExecCommand7 =<br /> #ExecCommand8 =<br /> #ExecCommand9 =<br /> #ExecCommand10 =</p> <p># Display the cursor in the line with the selected file. Some braille<br /> # readers (the Handy Tech modular series ZMU 737, for example) use the<br /> # cursor to focus and can make use of it to present the file line even<br /> # when other fields are changing.<br /> #UseCursorSelection = no</p> <p># Set the terminal title when running under xterm.<br /> #SetXtermTitle = yes</p> <p># Set the terminal title when running under screen.<br /> #SetScreenTitle = yes</p> <p># Display full paths instead of just file names in the playlist.<br /> PlaylistFullPaths = no</p> <p># The following setting describes how block markers are displayed in<br /> # the play time progress bar. Its value is a string of exactly three<br /> # characters. The first character is displayed in a position which<br /> # corresponds to the time marked as the start of a block and the last<br /> # character to the time marked as the end of the block. The middle<br /> # character is displayed instead if both the start and the end of the block<br /> # would fall in the same position (within the resolution of the interface).<br /> # You can turn off the displaying of these block marker positions by using<br /> # three space characters.<br /> #BlockDecorators = "`\"'"</p> <p># How long (in seconds) to leave a message displayed on the screen.<br /> # Setting this to a high value allows you to scroll through the messages<br /> # using the 'hide_message' key. Setting it to zero means you'll have to<br /> # be quick to see any message at all. Any new messages will be queued up<br /> # and displayed after the current message's linger time expires.<br /> #MessageLingerTime = 3</p> <p># Does MOC display a prefix on delayed messages indicating<br /> # the number of queued messages still to be displayed?<br /> #PrefixQueuedMessages = yes</p> <p># String to append to the queued message count if any<br /> # error messages are still waiting to be displayed.<br /> #ErrorMessagesQueued = "!"</p> <p># Self-describing ModPlug options (with 'yes' or 'no' values).<br /> #ModPlug_Oversampling = yes<br /> #ModPlug_NoiseReduction = yes<br /> #ModPlug_Reverb = no<br /> #ModPlug_MegaBass = no<br /> #ModPlug_Surround = no</p> <p># ModPlug resampling mode.<br /> # Valid values are:<br /> #<br /> # FIR - 8 tap fir filter (extremely high quality)<br /> # SPLINE - Cubic spline interpolation (high quality)<br /> # LINEAR - Linear interpolation (fast, good quality)<br /> # NEAREST - No interpolation (very fast, extremely bad sound quality)<br /> #<br /> #ModPlug_ResamplingMode = FIR</p> <p># Other self-describing ModPlug audio characteristic options.<br /> # (Note that the 32 bit sample size seems to be buggy.)<br /> #ModPlug_Channels = 2 # 1 or 2 channels<br /> #ModPlug_Bits = 16 # 8, 16 or 32 bits<br /> #ModPlug_Frequency = 44100 # 11025, 22050, 44100 or 48000 Hz<br /> #ModPlug_ReverbDepth = 0 # 0 (quiet) to 100 (loud)<br /> #ModPlug_ReverbDelay = 0 # Delay in ms (usually 40-200ms)<br /> #ModPlug_BassAmount = 0 # 0 (quiet) to 100 (loud).<br /> #ModPlug_BassRange = 10 # Cutoff in Hz (10-100).<br /> #ModPlug_SurroundDepth = 0 # Surround level 0(quiet)-100(heavy).<br /> #ModPlug_SurroundDelay = 0 # Surround delay in ms, usually 5-40ms.<br /> #ModPlug_LoopCount = 0 # 0 (never), n (times) or -1 (forever)</p> <p># Self-describing TiMidity audio characteristic options.<br /> #TiMidity_Rate = 44100 # Between 8000 and 48000<br /> #TiMidity_Bits = 16 # 8 or 16<br /> #TiMidity_Channels = 2 # 1 or 2<br /> #TiMidity_Volume = 100 # 0 to 800</p> <p># You can setup a TiMidity-Config-File here.<br /> # Leave it unset to use library defaults (/etc/timidity.cfg mostly).<br /> # Setting it to 'yes' also uses the library defaults.<br /> # Set it to 'no' if you don't have any configuration file.<br /> # Otherwise set it to the name of a specific file.<br /> #TiMidity_Config =</p> <p># Self-describing SidPlay2 audio characteristic options.<br /> #SidPlay2_DefaultSongLength = 180 # If not in database (in seconds)<br /> #SidPlay2_MinimumSongLength = 0 # Play at least n (in seconds)<br /> #SidPlay2_Frequency = 44100 # 4000 to 48000<br /> #SidPlay2_Bits = 16 # 8 or 16<br /> #SidPlay2_Optimisation = 0 # 0 (worst quality) to 2 (best quality)</p> <p># Set path to a HVSC-compatible database (if not set, database is disabled).<br /> #SidPlay2_Database =</p> <p># SidPlay2 playback Mode:<br /> #<br /> # "M": Mono (best for many SIDs)<br /> # "S": Stereo<br /> # "L"/"R": Left / Right<br /> #<br /> #SidPlay2_PlayMode = "M"</p> <p># Use start-song information from SID ('yes') or start at first song<br /> # ('no'). Songs before the start-song won't be played. (Note that this<br /> # option previously took the values 1 and 0; these are now deprecated<br /> # in favour of 'yes' and 'no'.)<br /> #SidPlay2_StartAtStart = yes</p> <p># Play sub-tunes. (Note that this option previously took the values 1<br /> # and 0; these are now deprecated in favour of 'yes' and 'no'.)<br /> #SidPlay2_PlaySubTunes = yes</p> <p># Run the OnSongChange command when a new song starts playing.<br /> # Specify the full path (i.e. no leading '~') of an executable to run.<br /> # Arguments will be passed, and you can use the following escapes:<br /> #<br /> # %a artist<br /> # %r album<br /> # %f filename<br /> # %t title<br /> # %n track<br /> # %d file duration in XX:YY form<br /> # %D file duration, number of seconds<br /> #<br /> # No pipes/redirects can be used directly, but writing a shell script<br /> # can do the job.<br /> #<br /> # Example: OnSongChange = "/home/jack/.moc/myscript %a %r"<br /> #<br /> #OnSongChange =</p> <p># If RepeatSongChange is 'yes' then MOC will execute the command every time<br /> # a song starts playing regardless of whether or not it is just repeating.<br /> # Otherwise the command will only be executed when a different song is<br /> # started.<br /> #RepeatSongChange = no</p> <p># Run the OnStop command (full path, no arguments) when MOC changes state<br /> # to stopped (i.e., when user stopped playing or changes a song).<br /> #OnStop = "/home/jack/.moc/myscript_on_stop"</p> <p># This option determines which song to play after finishing all the songs<br /> # in the queue. Setting this to 'yes' causes MOC to play the song which<br /> # follows the song being played before queue playing started. If set to<br /> # 'no', MOC will play the song following the last song in the queue if it<br /> # is in the playlist. The default is 'yes' because this is the way other<br /> # players usually behave. (Note that this option previously took the<br /> # values 1 and 0; these are now deprecated in favour of 'yes' and 'no'.)<br /> #QueueNextSongReturn = yes

Tried to active these ones, but i did not notice difference.
#PreferredDecoders += wav(sndfile,*,ffmpeg)<br /> PreferredDecoders += wv(wavpack,*,ffmpeg)<br /> PreferredDecoders += audio/wav(sndfile,*)

Please tell me what can i do under this configuration concretely.

Thank you for your message, and your support.

Kindly regards,
Nili

This line says you should be using the sndfile decoder (if you have it installed) first by default:

<br /> PreferredDecoders += wav(sndfile,*,ffmpeg)<br />

(Explicitly configuring the commented defaults doesn't achieve anything.)

Do you have it sndfile (which sndfile-info)? If so, do you have the MOC sndfile decoder installed (in whichever directory your particular distribution installs MOC plug-ins)?

PreferredDecoders += wav(sndfile)

'ldd /usr/lib/x86_64-linux-gnu/moc/decoder_plugins/libsndfile_decoder.so| grep libsnd'
libsndfile.so.1 => /usr/lib/x86_64-linux-gnu/libsndfile.so.1 (0x00007ff6c12dc000)

The bitrate isn't shown.

Elimar
--
Hates to do BBCode cosmetics

JCF must have been a bit sleepy while recommending the use of sndfile decoder to solve the issue ;)

static int sndfile_get_bitrate (void *unused ATTR_UNUSED)
{
return -1;
}

I don't see any way to get bitrate via libsndfile (unless I'm also missing something): http://www.mega-nerd.com/libsndfile/api.html However it could be possible to do the calculation at least for PCM .wav files. In the similar way, sndfile decoder should also try to better fit output sample format to the file contents - now it just converts everything to float.


JCF must have been a bit sleepy while recommending the use of sndfile decoder to solve the issue ;)

You'd be surprised how close to the mark that is.

No haven't installed sndfile, i'm just using default moc from Debian stable with plain minimal config.

Also no plugin have installed. Thanks for clarification, then is not it required to activate that settings since sndfile is not installed.