MADPLAY(1) MAD MADPLAY(1)
MPEG Audio Decoder MPEG Audio Decoder
03 April 2001
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 front-end to the MAD library.
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.
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.
OPTIONS
Verbosity
-v or --verbose
Show information about the stream during decoding, including
playing time, audio layer, bit rate, sampling frequency, and
stereo mode.
-q or --quiet
Do not show any information during decoding, except warnings.
-Q or --very-quiet
Do not show any information during decoding, including warnings.
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 below.) If a
format is not specified, one will be inferred from path. If path
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MADPLAY(1) MAD MADPLAY(1)
MPEG Audio Decoder MPEG Audio Decoder
03 April 2001
is a single dash (-), the output will be written to standard
output.
-d or --no-dither
Do not dither the output PCM samples. This may result in lower
sound quality but is useful for analyzing output from the
decoder.
--downsample
Reduce the output sampling frequency 2:1. This also reduces the
computational overhead of 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 decibel value must be in the range
-175 to +18. The value may be fractional, e.g. -1.5. A value of
0 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.
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 be 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.
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MPEG Audio Decoder MPEG Audio Decoder
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-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
still be stopped 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.
Miscellaneous
-V or --version
Display the effective version and build options for madplay, and
exit.
--license
Display a license message and exit.
-h or --help
Display a usage message and exit.
Output Formats
Other than playing on the native audio device, the following output
formats are supported:
raw binary signed 16-bit host-endian linear PCM, stereo interleaved
wave Microsoft RIFF/WAVE, 16-bit PCM format (*.wav)
snd Sun/NeXT audio file format, 8-bit ISDN mu-law (*.au, *.snd)
hex hexadecimal signed 24-bit linear PCM, stereo interleaved, one
sample per output line (debug builds only)
null no output (usually for testing or timing the decoder)
Time Specifications
For options which accept a time or duration argument, the following
time specifications are recognized:
hh:mm:ss.
Hours, minutes, seconds, and decimal fractions of a second. This
specification is flexible; hh:mm:ss, mmm:ss, :ss, sss.ddd, .ddd,
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03 April 2001
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 it will be used to
help restore the bit reservoir for the next frame.
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.
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
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MADPLAY(1) MAD MADPLAY(1)
MPEG Audio Decoder MPEG Audio Decoder
03 April 2001
most configurations, MAD is a Full Layer III ISO/IEC 11172-3 audio
decoder as defined by the standard.
NOTES
Because MAD produces output samples with a precision greater than 24
bits, by default madplay will dither the output to the necessary
number of bits, usually 16. 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.
MAD is implemented entirely without the use of floating point
arithmetic and should therefore perform well on all architectures,
especially those without an FPU.
BUGS
Granularity of start and stop times (--start and --time) is not yet as
fine as this specification suggests.
AUTHOR
Robert Leslie <rob@mars.org>
SEE ALSO
ISO/IEC 11172-3, ISO/IEC 13818-3; tests for compliance are defined in
ISO/IEC 11172-4 and ISO/IEC 13818-4.
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