Provided by: madplay_0.15.2b-8_amd64 bug

NAME

       madplay - decode and play MPEG audio stream(s)

SYNOPSIS

       madplay [options] file ...
       madplay [options] -o [type:]path file ...

DESCRIPTION

       madplay is a command-line MPEG audio decoder and player based on the MAD library (libmad).

       MAD  is  a  high-quality  MPEG  audio decoder. It currently supports MPEG-1 and the MPEG-2
       extension to Lower Sampling Frequencies, as well as the  so-called  MPEG 2.5  format.  All
       three audio layers (Layer I, Layer II, and Layer III a.k.a. MP3) are fully implemented.

       Among the special features of MAD are 24-bit PCM resolution and 100% fixed-point (integer)
       computation. Since  MAD  is  implemented  entirely  without  the  use  of  floating  point
       arithmetic, it performs especially well on architectures without an FPU.

       MAD  does  not  yet  support  MPEG-2  multichannel  audio  (although it should be backward
       compatible with such streams) nor does it currently support AAC.

       By default madplay reads and decodes one or more input files containing  MPEG  audio  data
       and plays them on the native audio device. If the input file is a single dash (-), data is
       read from standard input.

       Decoded output may optionally be redirected to a file instead of being played on the audio
       device by using the -o (--output) option.

       For  each  file,  madplay  will  also attempt to read and display ID3 tag information. The
       supported tag versions are ID3v1,  ID3v1.1,  ID3v2.2,  ID3v2.3,  and  ID3v2.4.  If  a  tag
       contains  relative  volume adjustment information (RVA2), madplay will use the information
       to adjust the master volume  for  output.  This  behavior  can  be  changed  with  the  -A
       (--adjust-volume) and -G (--replay-gain) options.

       If the -T (--show-tags-only) option is used, decoding is not performed but tag information
       is still displayed. When used in conjunction with -v (--verbose), encoder as well  as  ID3
       tags are shown.

OPTIONS

   Verbosity
       -v or --verbose
              Generally show more information than the default. During decoding, show information
              about the stream including playing time, audio layer, bit rate, sampling frequency,
              and stereo mode.

       -q or --quiet
              Generally  show  less  information  than  the  default. Do not show any information
              during decoding except warnings.

       -Q or --very-quiet
              Generally show no information except severe errors. Do not show any information  or
              warnings during decoding.

       --display-time=mode
              Set  the default verbose time display mode to mode, which must be one of remaining,
              current, or overall.  This is only relevant with -v (--verbose).  See --tty-control
              below for details on changing the time display mode during playback.

   Decoding
       --downsample
              Reduce  the  decoded  sampling  frequency  2:1. This also reduces the computational
              overhead of the decoder.

       -i or --ignore-crc
              Ignore CRC information in the audio stream. This causes frames with CRC  errors  to
              be  decoded  and  played  anyway.  This  option  is not recommended, but since some
              encoders have been known to generate bad CRC information, this option  is  a  work-
              around to play streams from such encoders.

       --ancillary-output=path
              Write  ancillary data from the MPEG audio stream to path.  If path is a single dash
              (-), the data will be written to standard output.  Bits  from  the  ancillary  data
              stream  are  packed into octets; if any bits remain, the final octet will be padded
              with zero bits. See the NOTES section below  for  further  information  about  this
              option.

   Audio Output
       -o or --output=[type:]path
              Direct  output  to  path, rather than playing audio on the native audio device. The
              format of the output is specified by type which can be any of the supported  output
              formats  (see  Output  Formats  below.)  If  a format is not specified, one will be
              inferred from path.  If path is a single dash (-), the output will  be  written  to
              standard output.

       -b or --bit-depth=depth
              Request  an  output  precision  of  depth  bits per sample. Higher bit depths yield
              higher quality sound. Typical bit depths are 8,  16,  24,  and  32,  however  other
              depths  may  also  be  possible.  Whether the request can be honored depends on the
              capabilities of the audio device or output format.  See the NOTES section below for
              further details about this option.

       -R or --sample-rate=hertz
              Request  an output sampling frequency of hertz samples per second (Hz).  The sample
              rate must be in the range 1000 to 65535 Hz.  Whether the  request  can  be  honored
              depends on the capabilities of the audio device or output format.  If the effective
              rate is not the same as the rate of the decoded audio,  output  may  be  resampled,
              possibly resulting in lower quality sound.

       -d or --no-dither
              Do  not  dither  output  PCM samples. This may result in lower quality sound but is
              useful for analyzing output from the decoder.

       --fade-in[=duration]
              Gradually fade-in the audio from each file over duration.  If  not  specified,  the
              default duration is 0:05 (five seconds.)

       -a or --attenuate=decibels or --amplify=decibels
              Attenuate  or amplify the signal by decibels (dB).  The signal is attenuated if the
              decibel value is negative; it is amplified if the value  is  positive.   The  value
              must be in the range -175 to +18 dB.  The value may be fractional, e.g. -1.5 dB.  A
              value  of  0 dB  will  leave  the  signal  unchanged.   Each  step  of  6 dB   will
              approximately  halve  (in  the  negative  direction)  or  double  (in  the positive
              direction) the strength of the signal.

       -A or --adjust-volume=decibels
              Adjust the relative volume for all files. This option overrides any per-file volume
              adjustment  settings.  For  example,  -A0  may  be  used  to ignore relative volume
              adjustments given by ID3 tags. Relative volume adjustments specified by this option
              or  by  ID3  tags  are  used as the base volume against which the signal is further
              attenuated or amplified using the -a (--attenuate, --amplify)  option  or  keyboard
              controls.  This option cannot be used together with -G (--replay-gain).

       -G or --replay-gain[=profile]
              Enable  Replay  Gain  volume  adjustments. Replay Gain information contained in the
              decoded files (if any) is used to make volume adjustments for output.  The  profile
              may  be  one of radio (the default) or audiophile.  See the NOTES section below for
              further details. When Replay Gain is enabled, a default pre-amp gain  of  +6 dB  is
              also applied; this can be changed with the -a (--attenuate, --amplify) option.

   Channel Selection
       For  dual  channel  streams, an output channel should be selected. If one is not selected,
       the first (left) channel will be used.

       For stereo streams, making a channel selection other than stereo will cause the output  to
       become monaural.

       -1 or --left
              Output the first (left) channel only.

       -2 or --right
              Output the second (right) channel only.

       -m or --mono
              Mix the left and right channels together.

       -S or --stereo
              Force stereo output, even if the stream is single or dual channel.

   Playback
       -s or --start=time
              Begin  playing  at  time,  given  as an offset from the beginning of the first file
              (0:00:00), seeking as necessary.

       -t or --time=duration
              Stop playback after the playing time of the output audio equals duration.

       -z or --shuffle
              Randomize the list of files given on the command line for playback.

       -r or --repeat[=max]
              Play  the  input  files  max  times,  or  indefinitely.  Playback  can  be  stopped
              prematurely  by  giving a time limit with the -t (--time) option. If -z (--shuffle)
              is also used, the files will be continuously shuffled and repeated in  such  a  way
              that  the same file is not played again until at least half of the other files have
              played in the interim.

       --tty-control
              Enable keyboard controls during playback. This is the default unless standard input
              is  not  a  terminal,  output  is  redirected  with  -o (--output), or either of -q
              (--quiet) or -Q (--very-quiet) is given.  The keyboard controls are:

              P  Pause; press any key to resume.

              S  Stop; press any key to replay the current file from the beginning.

              F  Forward; advance to the next file.

              B  Back; replay the current file, unless it  has  been  playing  for  less  than  4
                 seconds, in which case replay the previous file.

              T  Time display; change the time display mode. This only works with -v (--verbose).
                 The display mode alternates among overall playing time, current time  remaining,
                 and current playing time.

              +  Increase gain; increase the audio output gain by 0.5 dB.

              -  Decrease gain; decrease the audio output gain by 0.5 dB.

              Q  Quit; stop decoding and exit.

       --no-tty-control
              Disable  keyboard controls during playback. This is the default when standard input
              is not a terminal, output is  redirected  with  -o  (--output),  or  either  of  -q
              (--quiet) or -Q (--very-quiet) is given.

   Miscellaneous
       -T or --show-tags-only
              Show  ID3  and/or  encoder tags from the input files but do not otherwise decode or
              play any audio. By default only ID3 tags are shown (if any). With  -v  (--verbose),
              all  tags are shown. Encoder tags recognized by madplay include the Xing VBR header
              tag and the header tag format written by lame(1).

       -V or --version
              Display the effective version and build options for madplay and exit.

       --license
              Display copyright, license, and warranty information and exit.

       -h or --help
              Display usage information and exit.

Output Formats

       Other than playing on the native audio device, the following output formats are supported:

       cdda   CD audio, 16-bit big-endian 44100 Hz stereo PCM, padded to 2352-byte block boundary
              (*.cdr, *.cda)

       aiff   Audio IFF, [16-bit] PCM (*.aif, *.aiff)

       wave   Microsoft RIFF/WAVE, [16-bit] PCM (*.wav)

       snd    Sun/NeXT audio, 8-bit ISDN μ-law (*.au, *.snd)

       raw    binary [16-bit] host-endian linear PCM, stereo interleaved

       hex    ASCII  hexadecimal  [24-bit]  linear PCM, stereo interleaved, one sample per output
              line

       esd    Enlightened Sound Daemon (EsounD) [16-bit] (give speaker host as path)

       null   no output (usually for testing or timing the decoder)

       Default bit depths shown in square brackets can  be  changed  with  the  -b  (--bit-depth)
       option.

       Note that EsounD support requires the libesd library.

Time Specifications

       For  options  which  accept a time or duration argument, the following time specifications
       are recognized:

       hh:mm:ss.ddd
              Hours, minutes, seconds, and decimal fractions of a second. This  specification  is
              flexible;  hh:mm:ss,  mmm:ss,  :ss, sss.ddd, .ddd, and ssss are all acceptable. The
              component values are not constrained to any particular range or number of digits.

       frac/unit
              A length of time specified as a rational number, in seconds. This can be  used  for
              sample-granularity, for example 32/44100 for 32 samples, assuming a 44100 Hz sample
              frequency.

       time1+time2
              A composite time made by adding two time values together. This permits  mixing  the
              above specification forms.

       The resolution of any time value cannot exceed 1/352800000 seconds.

DIAGNOSTICS

       error: frame #: lost synchronization
              If  encountered  at the beginning of a file, this means the file contains something
              other than an ID3v2 tag before the MPEG audio data. If encountered in the middle of
              a file, it may mean the file is corrupt. This message is most commonly encountered,
              however, at the end of a file if the file contains an ID3v1 tag that is not aligned
              to  an  MPEG audio frame boundary. In this case, the message is harmless and may be
              ignored.

       error: frame #: bad main_data_begin pointer
              This message can occur while decoding a Layer III  stream  that  has  been  cut  or
              spliced without preserving its bit reservoir. The affected frame cannot be properly
              decoded, but will be used to help restore the bit reservoir for following frames.

       Most other messages indicate a deficiency in the input stream.

       When a frame cannot be properly decoded, a concealment strategy is used as follows:

       · If the previous frame was properly decoded, it is  repeated  in  place  of  the  current
         frame.

       · If the previous frame was not properly decoded, the current frame is muted.

NOTES

   Output Precision
       Because  MAD  produces  samples  with a precision greater than 24 bits, by default madplay
       will dither the samples to the precision of the output format. This produces high  quality
       audio  that  generally  sounds  superior  to  the  output  of a simple rounding algorithm.
       However, dithering may unfavorably affect an  analytic  examination  of  the  output,  and
       therefore it may be disabled by using the -d (--no-dither) option.

       The  actual precision of output samples can be requested with the -b (--bit-depth) option.
       Whether the request can be honored depends on the capabilities  of  the  audio  device  or
       output  format.  If  this  option  is  not specified, a typical default depth will be used
       (often 16) or in the case of output to an audio device, the highest bit  depth  determined
       to work reliably with the device will be used.

       Note  that bit depths greater than 24 are effectively the same as 24-bit precision samples
       padded to the requested depth.

   Ancillary Data
       MPEG audio streams contain an ancillary data stream in addition to audio data.  Most often
       this  does  not contain any useful information and may simply consist of padding bits. The
       MPEG-2 extension to multichannel audio uses  part  of  this  ancillary  stream  to  convey
       multichannel information; presently MAD does not interpret such data.

       For  applications which have uses for the stream, ancillary data can be extracted with the
       --ancillary-output option.

   Replay Gain
       madplay optionally supports the Replay Gain proposed standard with the -G  (--replay-gain)
       option  to  make compensating volume adjustments when playing decoded audio from different
       sources. There are two Replay Gain profiles: radio strives to make gain  adjustments  that
       give  all  tracks  equal  loudness,  while  audiophile  attempts  to  give ideal listening
       loudness. These adjustments are relative to a reference of 83 dB SPL.

       A pre-amp gain is also used in conjunction with Replay Gain to achieve the overall desired
       loudness. When Replay Gain is enabled, this pre-amp gain defaults to +6 dB, however it can
       be changed with the -a (--attenuate, --amplify) option or keyboard controls.

       Note that when enabled, Replay Gain overrides any relative volume adjustments specified by
       ID3  tags  (RVA2).  Replay Gain is also incompatible with the -A (--adjust-volume) option;
       any attempt to use it will be ignored.

       Replay Gain information is read either from an ID3 tag  (RGAD)  or  from  an  encoder  tag
       written by lame(1).  If both are present, the information in the ID3 tag takes precedence.
       In accordance with the proposed standard, if the requested  Replay  Gain  profile  is  not
       available but the alternate is, the alternate is used instead.

       Due  to  an  unfortunate  heresy,  versions  of  lame(1)  since  3.95.1  write Replay Gain
       information using a reference of 89 dB SPL instead of the 83 dB specified  in  the  Replay
       Gain  proposed  standard.  To  compensate,  madplay  automatically subtracts 6 dB from the
       Replay Gain values read from such tags.

       Note that madplay does not yet support hard limiting  as  suggested  by  the  Replay  Gain
       proposed standard; nor does it automatically reduce the pre-amp gain to avoid clipping.

CONFORMING TO

       MAD  conforms  to Part 3 of the ISO/IEC 11172 (MPEG-1) international standard for decoding
       MPEG audio. In addition, MAD supports the extension to Lower Sampling Frequencies (LSF) as
       defined in Part 3 of ISO/IEC 13818 (MPEG-2).

       The output from MAD has been tested and found to satisfy the ISO/IEC 11172-4 computational
       accuracy requirements for compliance. In most configurations,  MAD  is  a  Full  Layer III
       ISO/IEC 11172-3 audio decoder as defined by the standard.

       The ID3 tag parsing library used by madplay conforms to the ID3v2.4.0 informal standard.

       With  the exception of the clipping prevention provisions, Replay Gain support provided by
       madplay is in accordance with the Replay Gain proposed standard published on July 10, 2001
       by David Robinson.

BUGS

       The  resampling  algorithm  used by madplay is one of a linear interpolation, and does not
       produce optimum quality sound.

       The granularity of start and stop times (--start and --time) is not yet as  fine  as  this
       document suggests.

AUTHOR

       Robert Leslie <rob@mars.org>

SEE ALSO

       lame(1), normalize(1), sox(1), wget(1)