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Manpage of MADPLAY

MADPLAY

Section: MPEG Audio Decoder (1)
Updated: 02 November 2001
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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. If data is being read from standard input, ID3 tags will be displayed as they are encountered. The supported tag versions are ID3v1, ID3v1.1, ID3v2.2, ID3v2.3, and ID3v2.4.

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.

Decoding

--downsample: Reduce the output 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.

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. See the NOTES section below for further details about this option.
-d or --no-dither: Do not dither output PCM samples. This may result in lower sound quality 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 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 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 or output is redirected with -o (--output). 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.
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 or output is redirected with -o (--output).

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:

cdda: CD audio, 16-bit 44100 Hz stereo PCM (*.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 (debug builds only)
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 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 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.

NOTES

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 this request can be honored depends on the output format. If output is directed to an audio device, success depends on the capabilities of the device. 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.