flac/man/flac.md

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% flac(1) Version 1.4.2 | Free Lossless Audio Codec conversion tool
# NAME
flac - Free Lossless Audio Codec
# SYNOPSIS
**flac** \[ *OPTIONS* \] \[ *infile.wav* \| *infile.rf64* \|
*infile.aiff* \| *infile.raw* \| *infile.flac* \| *infile.oga* \|
*infile.ogg* \| **-** *...* \]
**flac** \[ **-d** \| **\--decode** \| **-t** \| **\--test** \| **-a** \|
**\--analyze** \] \[ *OPTIONS* \] \[ *infile.flac* \| *infile.oga* \|
*infile.ogg* \| **-** *...* \]
# DESCRIPTION
**flac** is a command-line tool for encoding, decoding, testing and
analyzing FLAC streams.
# GENERAL USAGE
**flac** supports as input RIFF WAVE, Wave64, RF64, AIFF, FLAC or Ogg
FLAC format, or raw interleaved samples. The decoder currently can output
to RIFF WAVE, Wave64, RF64, or AIFF format, or raw interleaved samples.
flac only supports linear PCM samples (in other words, no A-LAW, uLAW,
etc.), and the input must be between 4 and 32 bits per sample.
flac assumes that files ending in ".wav" or that have the RIFF WAVE
header present are WAVE files, files ending in ".w64" or have the Wave64
header present are Wave64 files, files ending in ".rf64" or have the
RF64 header present are RF64 files, files ending in ".aif" or ".aiff" or
have the AIFF header present are AIFF files, files ending in ".flac"
or have the FLAC header present are FLAC files and files ending in ".oga"
or ".ogg" or have the Ogg FLAC header present are Ogg FLAC files.
Other than this, flac makes no assumptions about file extensions, though
the convention is that FLAC files have the extension ".flac"
(or ".fla" on ancient "8.3" file systems like FAT-16).
Before going into the full command-line description, a few other things
help to sort it out:
1. flac encodes by default, so you must use -d to decode
2. the options -0 .. -8 (or --fast and --best) that control the
compression level actually are just synonyms for different groups of
specific encoding options (described later) and you can get the same
effect by using the same options. When specific options are specified
they take priority over the compression level no matter the order
3. flac behaves similarly to gzip in the way it handles input and output
files
4. the order in which options are specified is generally not important
Skip to the examples below for examples of some common tasks.
flac will be invoked one of four ways, depending on whether you are
encoding, decoding, testing, or analyzing. Encoding is the default
invocation, but can be switch to decoding with **-d**, analysis with
**-a** or testing with **-t**. Depending on which way is chosen,
encoding, decoding, analysis or testing options can be used, see section
OPTIONS for details. General options can be used for all.
If only one inputfile is specified, it may be "-" for stdin. When stdin
is used as input, flac will write to stdout. Otherwise flac will perform
the desired operation on each input file to similarly named output files
(meaning for encoding, the extension will be replaced with ".flac", or
appended with ".flac" if the input file has no extension, and for
decoding, the extension will be ".wav" for WAVE output and ".raw" for raw
output). The original file is not deleted unless --delete-input-file is
specified.
If you are encoding/decoding from stdin to a file, you should use the -o
option like so:
flac [options] -o outputfile
flac -d [options] -o outputfile
which are better than:
flac [options] > outputfile
flac -d [options] > outputfile
since the former allows flac to seek backwards to write the STREAMINFO or
RIFF WAVE header contents when necessary.
Also, you can force output data to go to stdout using -c.
To encode or decode files that start with a dash, use -- to signal the
end of options, to keep the filenames themselves from being treated as
options:
flac -V -- -01-filename.wav
The encoding options affect the compression ratio and encoding speed. The
format options are used to tell flac the arrangement of samples if the
input file (or output file when decoding) is a raw file. If it is a RIFF
WAVE, Wave64, RF64, or AIFF file the format options are not needed since
they are read from the file's header.
In test mode, flac acts just like in decode mode, except no output file
is written. Both decode and test modes detect errors in the stream, but
they also detect when the MD5 signature of the decoded audio does not
match the stored MD5 signature, even when the bitstream is valid.
flac can also re-encode FLAC files. In other words, you can specify a
FLAC or Ogg FLAC file as an input to the encoder and it will decoder it
and re-encode it according to the options you specify. It will also
preserve all the metadata unless you override it with other options (e.g.
specifying new tags, seekpoints, cuesheet, padding, etc.).
flac has been tuned so that the default settings yield a good speed vs.
compression tradeoff for many kinds of input. However, if you are looking
to maximize the compression rate or speed, or want to use the full power
of FLAC's metadata system, see the page titled 'About the FLAC Format' on
the FLAC website.
# EXAMPLES
Some common **encoding** tasks using flac:
`flac abc.wav`
: Encode abc.wav to abc.flac using the default compression setting. abc.wav is not deleted.
`flac --delete-input-file abc.wav`
: Like above, except abc.wav is deleted if there were no errors.
`flac --delete-input-file -w abc.wav`
: Like above, except abc.wav is deleted if there were no errors or warnings.
`flac --best abc.wav`
: Encode abc.wav to abc.flac using the highest compression setting.
`flac --verify abc.wav`
: Encode abc.wav to abc.flac and internally decode abc.flac to make sure it matches abc.wav.
`flac -o my.flac abc.wav`
: Encode abc.wav to my.flac.
`flac -T "TITLE=Bohemian Rhapsody" -T "ARTIST=Queen" abc.wav`
: Encode abc.wav and add some tags at the same time to abc.flac.
`flac *.wav`
: Encode all .wav files in the current directory.
`flac abc.aiff`
: Encode abc.aiff to abc.flac.
`flac abc.rf64`
: Encode abc.rf64 to abc.flac.
`flac abc.w64`
: Encode abc.w64 to abc.flac.
`flac abc.flac --force`
: This one's a little tricky: notice that flac is in encode mode by
default (you have to specify -d to decode) so this command actually
recompresses abc.flac back to abc.flac. --force is needed to make
sure you really want to overwrite abc.flac with a new version. Why
would you want to do this? It allows you to recompress an existing
FLAC file with (usually) higher compression options or a newer
version of FLAC and preserve all the metadata like tags too.
Some common **decoding** tasks using flac:
`flac -d abc.flac`
: Decode abc.flac to abc.wav. abc.flac is not deleted. NOTE: Without
-d it means re-encode abc.flac to abc.flac (see above).
`flac -d --force-aiff-format abc.flac`
`flac -d -o abc.aiff abc.flac`
: Two different ways of decoding abc.flac to abc.aiff (AIFF format).
abc.flac is not deleted.
`flac -d --force-rf64-format abc.flac`
`flac -d -o abc.rf64 abc.flac`
: Two different ways of decoding abc.flac to abc.rf64 (RF64 format).
abc.flac is not deleted.
`flac -d --force-wave64-format abc.flac`
`flac -d -o abc.w64 abc.flac`
: Two different ways of decoding abc.flac to abc.w64 (Wave64 format).
abc.flac is not deleted.
`flac -d -F abc.flac`
: Decode abc.flac to abc.wav and don't abort if errors are found
(useful for recovering as much as possible from corrupted files).
# OPTIONS
A summary of options is included below. For a complete description, see
the HTML documentation.
## GENERAL OPTIONS
**-v, \--version**
: Show the flac version number
**-h, \--help**
: Show basic usage and a list of all options
**-H, \--explain**
: Show detailed explanation of usage and all options
**-d, \--decode**
: Decode (the default behavior is to encode)
**-t, \--test**
: Test a flac encoded file (same as -d except no decoded file is written)
**-a, \--analyze**
: Analyze a FLAC encoded file (same as -d except an analysis file is
written)
**-c, \--stdout**
: Write output to stdout
**-s, \--silent**
: Silent mode (do not write runtime encode/decode statistics to stderr)
**\--totally-silent**
: Do not print anything of any kind, including warnings or errors. The
exit code will be the only way to determine successful completion.
**\--no-utf8-convert**
: Do not convert tags from local charset to UTF-8. This is useful for
scripts, and setting tags in situations where the locale is wrong.
This option must appear before any tag options!
**-w, \--warnings-as-errors**
: Treat all warnings as errors (which cause flac to terminate with a
non-zero exit code).
**-f, \--force**
: Force overwriting of output files. By default, flac warns that the
output file already exists and continues to the next file.
**-o** *filename***, \--output-name=***filename*
: Force the output file name (usually flac just changes the extension).
May only be used when encoding a single file. May not be used in
conjunction with \--output-prefix.
**\--output-prefix=***string*
: Prefix each output file name with the given string. This can be
useful for encoding or decoding files to a different directory. Make
sure if your string is a path name that it ends with a trailing \`/'
(slash).
**\--delete-input-file**
: Automatically delete the input file after a successful encode or
decode. If there was an error (including a verify error) the input
file is left intact.
**\--preserve-modtime**
: Output files have their timestamps/permissions set to match those of
their inputs (this is default). Use \--no-preserve-modtime to make
output files have the current time and default permissions.
**\--keep-foreign-metadata**
: If encoding, save WAVE, RF64, or AIFF non-audio chunks in FLAC
metadata. If decoding, restore any saved non-audio chunks from FLAC
metadata when writing the decoded file. Foreign metadata cannot be
transcoded, e.g. WAVE chunks saved in a FLAC file cannot be restored
when decoding to AIFF. Input and output must be regular files (not
stdin or stdout).
**\--keep-foreign-metadata-if-present**
: Like \--keep-foreign-metadata, but without throwing an error if
foreign metadata cannot be found or restored, instead printing a
warning.
**\--skip={***\#***\|***mm:ss.ss***}**
: Skip over the first number of samples of the input. This works for
both encoding and decoding, but not testing. The alternative form
mm:ss.ss can be used to specify minutes, seconds, and fractions of a
second.
**\--until={***\#***\|\[***+***\|***-***\]***mm:ss.ss***}**
: Stop at the given sample number for each input file. This works for
both encoding and decoding, but not testing. The given sample number
is not included in the decoded output. The alternative form mm:ss.ss
can be used to specify minutes, seconds, and fractions of a second.
If a \`+' (plus) sign is at the beginning, the \--until point is
relative to the \--skip point. If a \`-' (minus) sign is at the
beginning, the \--until point is relative to end of the audio.
**\--ogg**
: When encoding, generate Ogg FLAC output instead of native FLAC. Ogg
FLAC streams are FLAC streams wrapped in an Ogg transport layer. The
resulting file should have an '.oga' extension and will still be
decodable by flac. When decoding, force the input to be treated as
Ogg FLAC. This is useful when piping input from stdin or when the
filename does not end in '.oga' or '.ogg'.
**\--serial-number=***\#*
: When used with \--ogg, specifies the serial number to use for the
first Ogg FLAC stream, which is then incremented for each additional
stream. When encoding and no serial number is given, flac uses a
random number for the first stream, then increments it for each
additional stream. When decoding and no number is given, flac uses
the serial number of the first page.
## ANALYSIS OPTIONS
**\--residual-text**
: Includes the residual signal in the analysis file. This will make the
file very big, much larger than even the decoded file.
**\--residual-gnuplot**
: Generates a gnuplot file for every subframe; each file will contain
the residual distribution of the subframe. This will create a lot of
files.
## DECODING OPTIONS
**\--cue=\[***\#.#***\]\[-\[***\#.#***\]\]**
: Set the beginning and ending cuepoints to decode. The optional first
\#.# is the track and index point at which decoding will start; the
default is the beginning of the stream. The optional second \#.# is
the track and index point at which decoding will end; the default is
the end of the stream. If the cuepoint does not exist, the closest
one before it (for the start point) or after it (for the end point)
will be used. If those don't exist, the start of the stream (for the
start point) or end of the stream (for the end point) will be used.
The cuepoints are merely translated into sample numbers then used as
\--skip and \--until. A CD track can always be cued by, for example,
\--cue=9.1-10.1 for track 9, even if the CD has no 10th track.
**-F, \--decode-through-errors**
: By default flac stops decoding with an error and removes the
partially decoded file if it encounters a bitstream error. With -F,
errors are still printed but flac will continue decoding to
completion. Note that errors may cause the decoded audio to be
missing some samples or have silent sections.
**\--apply-replaygain-which-is-not-lossless\[=\<specification\>\]**
: Applies ReplayGain values while decoding. **WARNING: THIS IS NOT
LOSSLESS. DECODED AUDIO WILL NOT BE IDENTICAL TO THE ORIGINAL WITH
THIS OPTION.** This option is useful for example in transcoding
media servers, where the client does not support ReplayGain. For
details on the use of this option, see the section **ReplayGain
application specification**.
## ENCODING OPTIONS
**-V, \--verify**
: Verify a correct encoding by decoding the output in parallel and
comparing to the original
**\--lax**
: Allow encoder to generate non-Subset files. The resulting FLAC file
may not be streamable or might have trouble being played in all
players (especially hardware devices), so you should only use this
option in combination with custom encoding options meant for
archival.
**\--replay-gain**
: Calculate ReplayGain values and store them as FLAC tags, similar to
vorbisgain. Title gains/peaks will be computed for each input file,
and an album gain/peak will be computed for all files. All input
files must have the same resolution, sample rate, and number of
channels. Only mono and stereo files are allowed, and the sample
rate must be one of 8, 11.025, 12, 16, 22.05, 24, 32, 44.1, or 48
kHz. Also note that this option may leave a few extra bytes in a
PADDING block as the exact size of the tags is not known until all
files are processed. Note that this option cannot be used when
encoding to standard output (stdout).
**\--cuesheet=***filename*
: Import the given cuesheet file and store it in a CUESHEET metadata
block. This option may only be used when encoding a single file. A
seekpoint will be added for each index point in the cuesheet to the
SEEKTABLE unless \--no-cued-seekpoints is specified.
**\--picture={***FILENAME***\|***SPECIFICATION***}**
: Import a picture and store it in a PICTURE metadata block. More than
one \--picture option can be specified. Either a filename for the
picture file or a more complete specification form can be used. The
SPECIFICATION is a string whose parts are separated by \| (pipe)
characters. Some parts may be left empty to invoke default values.
FILENAME is just shorthand for "\|\|\|\|FILENAME". For the format of
SPECIFICATION, see the section **picture specification**.
**\--sector-align**
: Align encoding of multiple CD format files on sector boundaries. See the
HTML documentation for more information. This option is DEPRECATED and
may not exist in future versions of flac.
**\--ignore-chunk-sizes**
: When encoding to flac, ignore the file size headers in WAV and AIFF
files to attempt to work around problems with over-sized or malformed
files. WAV and AIFF files both have an unsigned 32 bit numbers in
the file header which specifes the length of audio data. Since this
number is unsigned 32 bits, that limits the size of a valid file to
being just over 4 Gigabytes. Files larger than this are mal-formed,
but should be read correctly using this option.
**-S {***\#***\|***X***\|***\#x***\|***\#s***}, \--seekpoint={***\#***\|***X***\|***\#x***\|***\#s***}**
: Include a point or points in a SEEKTABLE. Using \#, a seek point at
that sample number is added. Using X, a placeholder point is added at
the end of a the table. Using \#x, \# evenly spaced seek points will
be added, the first being at sample 0. Using \#s, a seekpoint will be
added every \# seconds (# does not have to be a whole number; it can
be, for example, 9.5, meaning a seekpoint every 9.5 seconds). You may
use many -S options; the resulting SEEKTABLE will be the unique-ified
union of all such values. With no -S options, flac defaults to
'-S 10s'. Use \--no-seektable for no SEEKTABLE. Note: '-S \#x' and
'-S \#s' will not work if the encoder can't determine the input size
before starting. Note: if you use '-S \#' and \# is \>= samples in
the input, there will be either no seek point entered (if the input
size is determinable before encoding starts) or a placeholder point
(if input size is not determinable).
**-P** *\#***, \--padding=***\#*
: Tell the encoder to write a PADDING metadata block of the given
length (in bytes) after the STREAMINFO block. This is useful if you
plan to tag the file later with an APPLICATION block; instead of
having to rewrite the entire file later just to insert your block,
you can write directly over the PADDING block. Note that the total
length of the PADDING block will be 4 bytes longer than the length
given because of the 4 metadata block header bytes. You can force no
PADDING block at all to be written with \--no-padding. The encoder
writes a PADDING block of 8192 bytes by default (or 65536 bytes if
the input audio stream is more that 20 minutes long).
**-T** *FIELD=VALUE***, \--tag=***FIELD=VALUE*
: Add a FLAC tag. The comment must adhere to the Vorbis comment spec;
i.e. the FIELD must contain only legal characters, terminated by an
'equals' sign. Make sure to quote the comment if necessary. This
option may appear more than once to add several comments. NOTE: all
tags will be added to all encoded files.
**\--tag-from-file=***FIELD=FILENAME*
: Like \--tag, except FILENAME is a file whose contents will be read
verbatim to set the tag value. The contents will be converted to
UTF-8 from the local charset. This can be used to store a cuesheet
in a tag (e.g. \--tag-from-file="CUESHEET=image.cue"). Do not try to
store binary data in tag fields! Use APPLICATION blocks for that.
**-b** *\#***, \--blocksize=***\#*
: Specify the blocksize in samples. The default is 1152 for -l 0,
else 4096. For subset streams this must be \<= 4608 if the samplerate
\<= 48kHz, for subset streams with higher samplerates it must be \<=
16384.
**-m, \--mid-side**
: Try mid-side coding for each frame (stereo input only)
**-M, \--adaptive-mid-side**
: Adaptive mid-side coding for all frames (stereo input only)
**-0..-8, \--compression-level-0..\--compression-level-8**
: Fastest compression..highest compression (default is -5). These are
synonyms for other options:
**-0, \--compression-level-0**
: Synonymous with -l 0 -b 1152 -r 3 \--no-mid-side
**-1, \--compression-level-1**
: Synonymous with -l 0 -b 1152 -M -r 3
**-2, \--compression-level-2**
: Synonymous with -l 0 -b 1152 -m -r 3
**-3, \--compression-level-3**
: Synonymous with -l 6 -b 4096 -r 4 \--no-mid-side
**-4, \--compression-level-4**
: Synonymous with -l 8 -b 4096 -M -r 4
**-5, \--compression-level-5**
: Synonymous with -l 8 -b 4096 -m -r 5
**-6, \--compression-level-6**
: Synonymous with -l 8 -b 4096 -m -r 6 -A subdivide_tukey(2)
**-7, \--compression-level-7**
: Synonymous with -l 12 -b 4096 -m -r 6 -A subdivide_tukey(2)
**-8, \--compression-level-8**
: Synonymous with -l 12 -b 4096 -m -r 6 -A subdivide_tukey(3)
**\--fast**
: Fastest compression. Currently synonymous with -0.
**\--best**
: Highest compression. Currently synonymous with -8.
**-e, \--exhaustive-model-search**
: Do exhaustive model search (expensive!)
**-A** *function***, \--apodization=***function*
: Window audio data with given the apodization function. See section
**Apodization functions** for details.
**-l** *\#***, \--max-lpc-order=***\#*
: Specifies the maximum LPC order. This number must be \<= 32. For
subset streams, it must be \<=12 if the sample rate is \<=48kHz. If
0, the encoder will not attempt generic linear prediction, and use
only fixed predictors. Using fixed predictors is faster but usually
results in files being 5-10% larger.
**-p, \--qlp-coeff-precision-search**
: Do exhaustive search of LP coefficient quantization (expensive!).
Overrides -q; does nothing if using -l 0
**-q** *\#***, \--qlp-coeff-precision=***\#*
: Precision of the quantized linear-predictor coefficients, 0 =\> let
encoder decide (min is 5, default is 0)
**-r \[***\#***,\]***\#***, \--rice-partition-order=\[***\#***,\]***\#*
: Set the \[min,\]max residual partition order (0..15). min defaults to
0 if unspecified. Default is -r 5.
## FORMAT OPTIONS
**\--endian={***big***\|***little***}**
: Set the byte order for samples
**\--channels=***\#*
: Set number of channels.
**\--bps=***\#*
: Set bits per sample.
**\--sample-rate=***\#*
: Set sample rate (in Hz).
**\--sign={***signed***\|***unsigned***}**
: Set the sign of samples (the default is signed).
**\--input-size=***\#*
: Specify the size of the raw input in bytes. If you are encoding raw
samples from stdin, you must set this option in order to be able to
use \--skip, \--until, \--cuesheet, or other options that need to
know the size of the input beforehand. If the size given is greater
than what is found in the input stream, the encoder will complain
about an unexpected end-of-file. If the size given is less, samples
will be truncated.
**\--force-raw-format**
: Force input (when encoding) or output (when decoding) to be treated
as raw samples (even if filename ends in *.wav*).
**\--force-aiff-format**
: Force the decoder to output AIFF format. This option is not needed if
the output filename (as set by -o) ends with *.aif* or *.aiff*. Also,
this option has no effect when encoding since input AIFF is
auto-detected.
**\--force-rf64-format**
: Force the decoder to output RF64 format. This option is not needed if
the output filename (as set by -o) ends with *.rf64*. Also, this
option has no effect when encoding since input RF64 is auto-detected.
**\--force-wave64-format**
: Force the decoder to output Wave64 format. This option is not needed
if the output filename (as set by -o) ends with *.w64*. Also, this
option has no effect when encoding since input Wave64 is
auto-detected.
## NEGATIVE OPTIONS
**\--no-adaptive-mid-side**
**\--no-cued-seekpoints**
**\--no-decode-through-errors**
**\--no-delete-input-file**
**\--no-preserve-modtime**
**\--no-keep-foreign-metadata**
**\--no-exhaustive-model-search**
**\--no-force**
**\--no-lax**
**\--no-mid-side**
**\--no-ogg**
**\--no-padding**
**\--no-qlp-coeff-prec-search**
**\--no-replay-gain**
**\--no-residual-gnuplot**
**\--no-residual-text**
**\--no-sector-align**
**\--no-seektable**
**\--no-silent**
**\--no-verify**
**\--no-warnings-as-errors**
These flags can be used to invert the sense of the corresponding normal
option.
## ReplayGain application specification
The option \--apply-replaygain-which-is-not-lossless\[=\<specification\>\]**
applies ReplayGain values while decoding. **WARNING: THIS IS NOT
LOSSLESS. DECODED AUDIO WILL NOT BE IDENTICAL TO THE ORIGINAL WITH THIS
OPTION.** This option is useful for example in transcoding media servers,
where the client does not support ReplayGain.
The equals sign and \<specification\> is optional. If omitted, the
default specification is 0aLn1.
The \<specification\> is a shorthand notation for describing how to apply
ReplayGain. All components are optional but order is important. '\[\]'
means 'optional'. '\|' means 'or'. '{}' means required. The format is:
\[\<preamp\>\]\[a\|t\]\[l\|L\]\[n{0\|1\|2\|3}\]
In which the following parameters are used:
- **preamp**: A floating point number in dB. This is added to the
existing gain value.
- **a\|t**: Specify 'a' to use the album gain, or 't' to use the track
gain. If tags for the preferred kind (album/track) do not exist but
tags for the other (track/album) do, those will be used instead.
- **l\|L**: Specify 'l' to peak-limit the output, so that the
ReplayGain peak value is full-scale. Specify 'L' to use a 6dB hard
limiter that kicks in when the signal approaches full-scale.
- **n{0\|1\|2\|3}**: Specify the amount of noise shaping. ReplayGain
synthesis happens in floating point; the result is dithered before
converting back to integer. This quantization adds noise. Noise
shaping tries to move the noise where you won't hear it as much.
0 means no noise shaping, 1 means 'low', 2 means 'medium', 3 means
'high'.
For example, the default of 0aLn1 means 0dB preamp, use album gain, 6dB
hard limit, low noise shaping. \--apply-replaygain-which-is-not-lossless=3
means 3dB preamp, use album gain, no limiting, no noise shaping.
flac uses the ReplayGain tags for the calculation. If a stream does
not have the required tags or they can't be parsed, decoding will
continue with a warning, and no ReplayGain is applied to that stream.
## Picture specification
This described the specification used for the **\--picture** option.
\[TYPE\]\|\[MIME-TYPE\]\|\[DESCRIPTION\]\|\[WIDTHxHEIGHTxDEPTH\[/COLORS\]\]\|FILE
TYPE is optional; it is a number from one of:
0. Other
1. 32x32 pixels 'file icon' (PNG only)
2. Other file icon
3. Cover (front)
4. Cover (back)
5. Leaflet page
6. Media (e.g. label side of CD)
7. Lead artist/lead performer/soloist
8. Artist/performer
9. Conductor
10. Band/Orchestra
11. Composer
12. Lyricist/text writer
13. Recording Location
14. During recording
15. During performance
16. Movie/video screen capture
17. A bright coloured fish
18. Illustration
19. Band/artist logotype
20. Publisher/Studio logotype
The default is 3 (front cover). There may only be one picture each of
type 1 and 2 in a file.
MIME-TYPE is optional; if left blank, it will be detected from the file.
For best compatibility with players, use pictures with MIME type
image/jpeg or image/png. The MIME type can also be \--\> to mean that
FILE is actually a URL to an image, though this use is discouraged.
DESCRIPTION is optional; the default is an empty string.
The next part specifies the resolution and color information. If the
MIME-TYPE is image/jpeg, image/png, or image/gif, you can usually leave
this empty and they can be detected from the file. Otherwise, you must
specify the width in pixels, height in pixels, and color depth in
bits-per-pixel. If the image has indexed colors you should also specify
the number of colors used. When manually specified, it is not checked
against the file for accuracy.
FILE is the path to the picture file to be imported, or the URL if MIME
type is \--\>
For example, "\|image/jpeg\|\|\|../cover.jpg" will embed the JPEG file
at ../cover.jpg, defaulting to type 3 (front cover) and an empty
description. The resolution and color info will be retrieved from the
file itself.
The specification
"4\|\--\>\|CD\|320x300x24/173\|http://blah.blah/backcover.tiff" will
embed the given URL, with type 4 (back cover), description "CD", and a
manually specified resolution of 320x300, 24 bits-per-pixel, and 173
colors. The file at the URL will not be fetched; the URL itself is
stored in the PICTURE metadata block.
## Apodization functions
To improve LPC analysis, audio data is windowed . The window can be
selected with one or more **-A** options. Possible functions are:
bartlett, bartlett_hann, blackman, blackman_harris_4term_92db,
connes, flattop, gauss(STDDEV), hamming, hann, kaiser_bessel, nuttall,
rectangle, triangle, tukey(P), partial_tukey(n\[/ov\[/P\]\]),
punchout_tukey(n\[/ov\[/P\]\]), subdivide_tukey(n\[/P\]) welch.
- For gauss(STDDEV), STDDEV is the standard deviation (0\<STDDEV\<=0.5).
- For tukey(P), P specifies the fraction of the window that is tapered
(0\<=P\<=1; P=0 corresponds to "rectangle" and P=1 corresponds to
"hann").
- For partial_tukey(n) and punchout_tukey(n), n apodization functions are
added that span different parts of each block. Values of 2 to 6 seem to
yield sane results. If necessary, an overlap can be specified, as can be
the taper parameter, for example partial_tukey(2/0.2) or
partial_tukey(2/0.2/0.5). ov should be smaller than 1 and can be
negative. The use of this is that different parts of a block are ignored
as the might contain transients which are hard to predict anyway. The
encoder will try each different added apodization (each covering a
different part of the block) to see which resulting predictor results in
the smallest representation.
- subdivide_tukey(n) is a more efficient reimplementation of partial_tukey
and punchout_tukey taken together, recycling as much data as possible. It
combines all possible non-redundant partial_tukey(n) and punchout_tukey(n)
up to the n specified. Specifying subdivide_tukey(3) is equivalent to
specifying tukey, partial_tukey(2), partial_tukey(3) and punchout_tukey(3),
specifying subdivide_tukey(5) equivalently adds partial_tukey(4),
punchout_tukey(4), partial_tukey(5) and punchout_tukey(5). To be able to
reuse data as much as possible, the tukey taper is taken equal for all
windows, and the P specified is applied for the smallest used window.
In other words, subdivide_tukey(2/0.5) results in a taper equal to that
of tukey(0.25) and subdivide_tukey(5) in a taper equal to that of
tukey(0.1). The default P for subdivide_tukey when none is specified is
0.5.
Note that P, STDDEV and ov are locale specific, so a comma as
decimal separator might be required instead of a dot. Use scientific
notation for a locale-independent specification, for example
tukey(5e-1) instead of tukey(0.5) or tukey(0,5).
More than one -A option (up to 32) may be used. Any function that is
specified erroneously is silently dropped. The encoder chooses suitable
defaults in the absence of any -A options; any -A option specified
replaces the default(s).
When more than one function is specified, then for every subframe the
encoder will try each of them separately and choose the window that
results in the smallest compressed subframe. Multiple functions can
greatly increase the encoding time.
# SEE ALSO
**metaflac(1)**
# AUTHOR
This manual page was initially written by Matt Zimmerman
\<mdz@debian.org\> for the Debian GNU/Linux system (but may be used by
others). It has been kept up-to-date by the Xiph.org Foundation.