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@D$,@EΐUSd$xstxs[Ћud$[]ÐUS[ÈW{Y[fLaCOggSFLAC?%s%s
Usage: oggenc [options] inputfile [...]

OPTIONS:
 General:
 -Q, --quiet          Produce no output to stderr
 -h, --help           Print this help text
 -v, --version        Print the version number
 -k, --skeleton       Adds an Ogg Skeleton bitstream
 -r, --raw            Raw mode. Input files are read directly as PCM data
 -B, --raw-bits=n     Set bits/sample for raw input. Default is 16
 -C, --raw-chan=n     Set number of channels for raw input. Default is 2
 -R, --raw-rate=n     Set samples/sec for raw input. Default is 44100
 --raw-endianness     1 for bigendian, 0 for little (defaults to 0)
 -b, --bitrate        Choose a nominal bitrate to encode at. Attempt
                      to encode at a bitrate averaging this. Takes an
                      argument in kbps. By default, this produces a VBR
                      encoding, equivalent to using -q or --quality.
                      See the --managed option to use a managed bitrate
                      targetting the selected bitrate.
 --managed            Enable the bitrate management engine. This will allow
                      much greater control over the precise bitrate(s) used,
                      but encoding will be much slower. Don't use it unless
                      you have a strong need for detailed control over
                      bitrate, such as for streaming.
 -m, --min-bitrate    Specify a minimum bitrate (in kbps). Useful for
                      encoding for a fixed-size channel. Using this will
                      automatically enable managed bitrate mode (see
                      --managed).
 -M, --max-bitrate    Specify a maximum bitrate in kbps. Useful for
                      streaming applications. Using this will automatically
                      enable managed bitrate mode (see --managed).
 --advanced-encode-option option=value
                      Sets an advanced encoder option to the given value.
                      The valid options (and their values) are documented
                      in the man page supplied with this program. They are
                      for advanced users only, and should be used with
                      caution.
 -q, --quality        Specify quality, between -1 (very low) and 10 (very
                      high), instead of specifying a particular bitrate.
                      This is the normal mode of operation.
                      Fractional qualities (e.g. 2.75) are permitted
                      The default quality level is 3.
 --resample n         Resample input data to sampling rate n (Hz)
 --downmix            Downmix stereo to mono. Only allowed on stereo
                      input.
 -s, --serial         Specify a serial number for the stream. If encoding
                      multiple files, this will be incremented for each
                      stream after the first.
 --discard-comments   Prevents comments in FLAC and Ogg FLAC files from
                      being copied to the output Ogg Vorbis file.

 Naming:
 -o, --output=fn      Write file to fn (only valid in single-file mode)
 -n, --names=string   Produce filenames as this string, with %%a, %%t, %%l,
                      %%n, %%d replaced by artist, title, album, track number,
                      and date, respectively (see below for specifying these).
                      %%%% gives a literal %%.
 -X, --name-remove=s  Remove the specified characters from parameters to the
                      -n format string. Useful to ensure legal filenames.
 -P, --name-replace=s Replace characters removed by --name-remove with the
                      characters specified. If this string is shorter than the
                      --name-remove list or is not specified, the extra
                      characters are just removed.
                      Default settings for the above two arguments are platform
                      specific.
 -c, --comment=c      Add the given string as an extra comment. This may be
                      used multiple times. The argument should be in the
                      format "tag=value".
 -d, --date           Date for track (usually date of performance)
 -N, --tracknum       Track number for this track
 -t, --title          Title for this track
 -l, --album          Name of album
 -a, --artist         Name of artist
 -G, --genre          Genre of track
                      If multiple input files are given, then multiple
                      instances of the previous five arguments will be used,
                      in the order they are given. If fewer titles are
                      specified than files, OggEnc will print a warning, and
                      reuse the final one for the remaining files. If fewer
                      track numbers are given, the remaining files will be
                      unnumbered. For the others, the final tag will be reused
                      for all others without warning (so you can specify a date
                      once, for example, and have it used for all the files)

INPUT FILES:
 OggEnc input files must currently be 24, 16, or 8 bit PCM WAV, AIFF, or AIFF/C
 files, 32 bit IEEE floating point WAV, and optionally FLAC or Ogg FLAC. Files
  may be mono or stereo (or more channels) and any sample rate.
 Alternatively, the --raw option may be used to use a raw PCM data file, which
 must be 16 bit stereo little-endian PCM ('headerless wav'), unless additional
 parameters for raw mode are specified.
 You can specify taking the file from stdin by using - as the input filename.
 In this mode, output is to stdout unless an output filename is specified
 with -o

(c) 2000-2005 Michael Smith <msmith@xiph.org>
oggenc from vorbis-tools 1.2.0 
Couldn't convert comment to UTF-8, cannot add
WARNING: Insufficient titles specified, defaulting to final title.
WARNING: Ignoring illegal escape character '%c' in name format
Enabling bitrate management engine
WARNING: Raw endianness specified for non-raw data. Assuming input is raw.
WARNING: Couldn't read endianness argument "%s"
WARNING: Couldn't read resampling frequency "%s"
Warning: Resample rate specified as %d Hz. Did you mean %d Hz?
Warning: Couldn't parse scaling factor "%s"
No value for advanced encoder option found
Internal error parsing command line options
Warning: Illegal comment used ("%s"), ignoring.
Warning: nominal bitrate "%s" not recognised
Warning: minimum bitrate "%s" not recognised
Warning: maximum bitrate "%s" not recognised
Quality option "%s" not recognised, ignoring
WARNING: quality setting too high, setting to maximum quality.
WARNING: Multiple name formats specified, using final
WARNING: Multiple name format filters specified, using final
WARNING: Multiple name format filter replacements specified, using final
WARNING: Multiple output files specified, suggest using -n
WARNING: Raw bits/sample specified for non-raw data. Assuming input is raw.
WARNING: Invalid bits/sample specified, assuming 16.
WARNING: Raw channel count specified for non-raw data. Assuming input is raw.
WARNING: Invalid channel count specified, assuming 2.
WARNING: Raw sample rate specified for non-raw data. Assuming input is raw.
WARNING: Invalid sample rate specified, assuming 44100.
WARNING: Unknown option specified, ignoring->
A:a:b:B:c:C:d:G:hkl:m:M:n:N:o:P:q:QrR:s:t:vX:%s%s
ERROR: No input files specified. Use -h for help.
ERROR: Multiple files specified when using stdin
ERROR: Multiple input files with specified output filename: suggest using -n
ERROR: Cannot open input file "%s": %s
ERROR: Input file "%s" is not a supported format
WARNING: No filename, defaulting to "default.ogg"
ERROR: Could not create required subdirectories for output filename "%s"
ERROR: Input filename is the same as output filename "%s"
ERROR: Cannot open output file "%s": %s
Resampling input from %d Hz to %d Hz
WARNING: Can't downmix except from stereo to mono
titleartistgenredatealbumtracknumber(none)//usr/share/localevorbis-toolsskletonmanagedraw-endianness%dresampledownmixscale%fadvanced-encode-optiondiscard-comments%u-rbrawRAW file readerOpening with %s module: %s
(stdin).ogadefault.oggwbDownmixing stereo to mono
Scaling input to %f
quiethelpskeletoncommentnamesname-removename-replaceversionraw-bitsraw-chanraw-ratemin-bitratemax-bitratequalitytracknumserialddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddpdddddddddddP@ddddd8dddddddd8ddddddXPdxh(d?Internal error! Please report this bug.
Couldn't initialise resampler
BUG: Got zero samples from resampler: your file will be truncated. Please report this.
Big endian 24 bit PCM data is not currently supported, aborting.
Internal error: attempt to read unsupported bitdepth %d
Warning: Unexpected EOF in reading WAV header
Skipping chunk of type "%s", length %d
Warning: Unexpected EOF in AIFF chunk
Warning: No common chunk found in AIFF file
Warning: Truncated common chunk in AIFF header
Warning: Unexpected EOF in reading AIFF header
Warning: AIFF-C header truncated.
Warning: Can't handle compressed AIFF-C (%c%c%c%c)
Warning: No SSND chunk found in AIFF file
Warning: Corrupted SSND chunk in AIFF header
Warning: Unexpected EOF reading AIFF header
Warning: OggEnc does not support this type of AIFF/AIFC file
 Must be 8 or 16 bit PCM.
Warning: Unrecognised format chunk in WAV header
Warning: INVALID format chunk in wav header.
 Trying to read anyway (may not work)...
ERROR: Wav file is unsupported type (must be standard PCM
 or type 3 floating point PCM)
Warning: WAV 'block alignment' value is incorrect, ignoring.
The software that created this file is incorrect.
ERROR: Wav file is unsupported subformat (must be 8,16, or 24 bit PCM
or floating point PCM)
RIFFWAVECOMMNONEsowtSSNDFORMAIFfmt datawavWAV file readeraiffAIFF/AIFC file readerflacOgg FLAC file readerB?<84(min %d kbps, max %d kbps)(min %d kbps, no max)(no min, max %d kbps)(no min or max)standard outputstandard input
%s


Done encoding file "%s"


Done encoding.

	File length:  %dm %04.1fs
	Elapsed time: %dm %04.1fs
	Rate:         %.4f
	Average bitrate: %.1f kb/s


audio/vorbisContent-Typebitrate_average_damping%lfbitrate_average%ldbit_reservoir_biasbit_reservoir_bitsbitrate_hard_minbitrate_hard_maximpulse_noisetunelowpass_frequency|/-\Encoding %s%s%s to 
         %s%s%s 
at average bitrate %d kbps , 
using full bitrate management engine
Encoding %s%s%s to 
         %s%s%s 
at approximate bitrate %d kbps (VBR encoding enabled)
Encoding %s%s%s to 
         %s%s%s 
at quality level %2.2f using constrained VBR Encoding %s%s%s to 
         %s%s%s 
at quality %2.2f
Encoding %s%s%s to 
         %s%s%s 
using bitrate management 	Encoding [%2dm%.2ds so far] %c 	[%5.1f%%] [%2dm%.2ds remaining] %c 255 channels should be enough for anyone. (Sorry, but Vorbis doesn't support more)
Requesting a minimum or maximum bitrate requires --managed
Mode initialisation failed: invalid parameters for quality
Set optional hard quality restrictions
Failed to set bitrate min/max in quality mode
Mode initialisation failed: invalid parameters for bitrate
Setting advanced encoder option "%s" to %s
For option %s, couldn't read value %s as double
For option %s, couldn't read value %s as integer
Changed lowpass frequency from %f kHz to %f kHz
Unrecognised advanced option "%s"
Failed to set advanced rate management parameters
Failed writing fishead packet to output stream
Failed writing header to output stream
Failed writing fisbone header packet to output stream
Failed writing skeleton eos packet to output stream
Failed writing data to output stream
GG ApB??Mb?Couldn't create directory "%s": %s
Error checking for existence of directory %s: %s
Error: path segment "%s" is not a directory
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