flac - Free Lossless Audio Codec

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.

Version 1.5.0                                                                                            flac(1)

Description

flac is a command-line tool for encoding, decoding, testing and analyzing FLAC streams.

Examples

Some typical encoding and decoding tasks using flac:

Encodingexamples
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 and no warnings.

flac --best abc.wav or flac -8 abc.wav
Encode abc.wav to abc.flac using the highest compression preset.

flac --verify abc.wav or flac -V 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 abc.aiff foo.rf64 bar.w64
Encode abc.aiff to abc.flac, foo.rf64 to foo.flac and bar.w64 to bar.flac

flac *.wav *.aif?
Wildcards are supported. This command will encode all .wav files and all .aif/.aiff/.aifc files
(as well as other supported files ending in .aif+one character) in the current directory.

flac abc.flac --force or flac abc.flac -f
Recompresses, keeping metadata like tags. The syntax is a little tricky: this is an encoding
command (which is the default: you need to specify -d for decoded output), and will thus want to
output the file abc.flac - which already exists. flac will require the --force or shortform -f
option to overwrite an existing file. Recompression will first write a temporary file, which
afterwards replaces the old abc.flac (provided flac has write access to that file). The above
example uses default settings. More often, recompression is combined with a different - usually
higher - compression option. Note: If the FLAC file does not end with .flac - say, it is abc.fla
- the -f is not needed: A new abc.flac will be created and the old kept, just like for an
uncompressed input file.

flac --tag-from-file="ALBUM=albumtitle.txt" -T "ARTIST=Queen" *.wav
Encode every .wav file in the directory and add some tags. Every file will get the same set of
tags. Warning: Will wipe all existing tags, when the input file is (Ogg) FLAC - not just those
tags listed in the option. Use the metaflac utility to tag FLAC files.

flac --keep-foreign-metadata-if-present abc.wav
FLAC files can store non-audio chunks of input WAVE/AIFF/RF64/W64 files. The related option
--keep-foreign-metadata works the same way, but will instead exit with an error if the input has
no such non-audio chunks. The encoder only stores the chunks as they are, it cannot import the
content into its own tags (vorbis comments). To transfer such tags from a source file, use
tagging software which supports them.

flac -Vj2 -m3fo Track07.flac -- -7.wav
flac employs the commonplace convention that options in a short version - invoked with single dash
- can be shortened together until one that takes an argument. Here -j and -o do, and after the
“2” a whitespace is needed to start new options with single/double dash. The -m option does not,
and the following “3” is the -3 compression setting. The options could equally well have been
written out as -V -j 2 -m -3 -f -o Track04.flac , or as -fo Track04.flac -3mVj2. flac also
employs the convention that -- (with whitespace!) signifies end of options, treating everything
to follow as filename. That is needed when an input filenames could otherwise be read as an
option, and “-7” is one such. In total, this line takes the input file -7.wav as input; -o will
give output filename as Track07.flac, and the -f will overwrite if the file Track04.flac is
already present. The encoder will select encoding preset -3 modified with the -m switch, and use
two CPU threads. Afterwards, the -V will make it decode the flac file and compare the audio to
the input, to ensure they are indeed equal.

Decodingexamples
flac --decode abc.flac or flac -d abc.flac
Decode abc.flac to abc.wav. abc.flac is not deleted. If abc.wav is already present, the process
will exit with an error instead of overwriting; use –force / -f to force overwrite. NOTE: A mere
flac abc.flac without–decodeoritsshortform-d, would mean to re-encode abc.flac to abc.flac
(see above), and that command would err out because abc.flac already exists.

flac -d --force-aiff-format abc.flac or flac -d -o abc.aiff abc.flac
Two different ways of decoding abc.flac to abc.aiff (AIFF format). abc.flac is not deleted. -d
-o could be shortened to -do. The decoder can force other output formats, or different versions
of the WAVE/AIFF formats, see the options below.

flac -d --keep-foreign-metadata-if-present abc.flac
If the FLAC file has non-audio chunks stored from the original input file, this option will
restore both audio and non-audio. The chunks will reveal the original file type, and the decoder
will select output format and output file extension accordingly - note that this is not compatible
with forcing a particular output format except if it coincides with the original, as the decoder
cannot transcode non-audio between formats. If there are no such chunks stored, it will decode to
abc.wav. The related option --keep-foreign-metadata will instead exit with an error if no such
non-audio chunks are found.

flac -d -F abc.flac
Decode abc.flac to abc.wav and don’t abort if errors are found. This is potentially useful for
recovering as much as possible from a corrupted file. Note: Be careful about trying to “repair”
files this way. Often it will only conceal an error, and not play any subjectively “better” than
the corrupted file. It is a good idea to at least keep it, and possibly try several decoders,
including the one that generated the file, and hear if one has less detrimental audible errors
than another. Make sure output volume is limited, as corrupted audio can generate loud noises.

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. Encoding 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).
3. The order in which options are specified is generally not important except when they contradict each
other, then the latter takes precedence except that compression presets are overridden by any option
given before or after. For example, -0M, -M0, -M2 and -2M are all the same as -1, and -l 12 -6 the
same as -7.

4. flac behaves similarly to gzip in the way it handles input and output files

Skip to the EXAMPLES section below for examples of some typical 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.

Name

       flac - Free Lossless Audio Codec

Options

A summary of options is included below. Several of the options can be negated, see the Negativeoptions
section below.

GENERALOPTIONS-v, --version
Show the flac version number, and quit.

-h, --help
Show basic usage and a list of all options, and quit.

-d, --decode
Decode (the default behavior is to encode)

-t, --test
Test a flac encoded file. This works the same as -d except no decoded file is written, and with
some additional checks like parsing of all metadata blocks.

-a, --analyze
Analyze a FLAC encoded file. This works the same as -d except the output is an analysis file, not
a decoded file.

-c, --stdout
Write output to stdout

-f, --force
Force overwriting of output files. By default, flac warns that the output file already exists and
continues to the next file.

--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.

-oFILENAME, --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).

--preserve-modtime
(Enabled by default.) Output files have their timestamps/permissions set to match those of their
inputs. 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). With this option, FLAC will pick
the right output format on decoding. It will exit with error if no such chunks are found.

--keep-foreign-metadata-if-present
Like --keep-foreign-metadata, but without throwing an error if foreign metadata cannot be found or
restored. Instead, prints a warning.

--skip={#|MM:SS}
Skip the first number of samples of the input. To skip over a given initial time, specify instead
minutes and seconds: there must then be at least one digit on each side of the colon sign.
Fractions of a second can be specified, with locale-dependent decimal point, e.g.
--skip=123:9,867 if your decimal point is a comma. A --skip option is applied to each input file
if more are given. This option cannot be used with -t. When used with -a, the analysis file will
enumerate frames from starting point.

--until={#|[+|]MM:SS}
Stop at the given sample number (which is not included). A negative number is taken relative to
the end of the audio, a `+’ (plus) sign means that the --until point is taken relative to the
--skip point. For other considerations, see --skip.

--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!

-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.

-w, --warnings-as-errors
Treat all warnings as errors (which cause flac to terminate with a non-zero exit code).

DECODINGOPTIONS-F, --decode-through-errors
By default flac stops decoding with an error message 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.

--cue=[#.#][-[#.#]]
Set the beginning and ending cuepoints to decode. Decimal points are locale-dependent (dot or
comma). 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.

–decode-chained-stream
Decode all links in a chained Ogg stream, not just the first one.

--apply-replaygain-which-is-not-lossless[=SPECIFICATION]
Applies ReplayGain values while decoding. WARNING:THISISNOTLOSSLESS.DECODEDAUDIOWILLNOTBEIDENTICALTOTHEORIGINALWITHTHISOPTION. 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 ReplayGainapplicationspecification.

ENCODINGOPTIONS
Encoding will default to -5, -A “tukey(5e-1)” and one CPU thread.

-V, --verify
Verify a correct encoding by decoding the output in parallel and comparing to the original.

-0, --compression-level-0, --fast
Fastest compression preset. Currently synonymous with -l 0 -b 1152 -r 3 --no-mid-side

-1, --compression-level-1
Currently synonymous with -l 0 -b 1152 -M -r 3

-2, --compression-level-2
Currently synonymous with -l 0 -b 1152 -m -r 3

-3, --compression-level-3
Currently synonymous with -l 6 -b 4096 -r 4 --no-mid-side

-4, --compression-level-4
Currently synonymous with -l 8 -b 4096 -M -r 4

-5, --compression-level-5
Currently synonymous with -l 8 -b 4096 -m -r 5

-6, --compression-level-6
Currently synonymous with -l 8 -b 4096 -m -r 6 -A "subdivide_tukey(2)"

-7, --compression-level-7
Currently synonymous with -l 12 -b 4096 -m -r 6 -A "subdivide_tukey(2)"

-8, --compression-level-8, --best
Currently synonymous with -l 12 -b 4096 -m -r 6 -A "subdivide_tukey(3)"

-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
only choose among a set of fixed (hard-coded) predictors. Restricting to fixed predictors only is
faster, but compresses weaker - typically five percentage points / ten percent larger files.

-b #, --blocksize=#
Specify the blocksize in samples. The current default is 1152 for -l 0, else 4096. Blocksize
must be between 16 and 65535 (inclusive). For subset streams it must be <= 4608 if the samplerate
is <= 48kHz, for subset streams with higher samplerates it must be <= 16384.

-m, --mid-side
Try mid-side coding for each frame (stereo only, otherwise ignored).

-M, --adaptive-mid-side
Adaptive mid-side coding for all frames (stereo only, otherwise ignored).

-r [#,]#, --rice-partition-order=[#,]#
Set the [min,]max residual partition order (0..15). For subset streams, max must be <=8. min
defaults to 0. Default is -r 5. Actual partitioning will be restricted by block size and
prediction order, and the encoder will silently reduce too high values.

-AFUNCTION(S), --apodization=FUNCTION(S)
Window audio data with given apodization function. More can be given, comma-separated. See
section Apodizationfunctions for details.

-e, --exhaustive-model-search
Do exhaustive model search (expensive!).

-q #, --qlp-coeff-precision=#
Precision of the quantized linear-predictor coefficients. This number must be in between 5 and
16, or 0 (the default) to let encoder decide. Does nothing if using -l 0.

-p, --qlp-coeff-precision-search
Do exhaustive search of LP coefficient quantization (expensive!). Overrides -q; does nothing if
using -l 0.

--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.

--limit-min-bitrate
Limit minimum bitrate by not allowing frames consisting of only constant subframes. This ensures
a bitrate of at least 1 bit/sample, for example 48kbit/s for 48kHz input. This is mainly useful
for internet streaming.

-j #, --threads=#
Try to set a maximum number of threads to use for encoding. If multithreading was not enabled on
compilation or when setting a number of threads that is too high, this fails with a warning. The
value of 0 means a default set by the encoder; currently that is 1 thread (i.e. no
multithreading), but that could change in the future. Currently, up to 128 threads are supported.
Using a value higher than the number of available CPU threads harms performance.

--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 specifies length of
audio data with an unsigned 32-bit number, limiting audio to just over 4 gigabytes. Files larger
than this are malformed, but should be read correctly using this option. Beware however, it could
misinterpret any data following the audio chunk, as audio.

--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 8, 11.025, 12, 16, 18.9, 22.05, 24, 28, 32,
36, 37.8, 44.1, 48, 56, 64, 72, 75.6, 88.2, 96, 112, 128, 144, 151.2, 176.4, 192, 224, 256, 288,
302.4, 352.8, 384, 448, 512, 576, or 604.8 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. Specifying only FILENAME is just shorthand for
“||||FILENAME”. See the section Picturespecification for SPECIFICATION format.

-S {#|X|#x|#s}, --seekpoint={#|X|#x|#s}
Specifies point(s) to include in SEEKTABLE, to override the encoder’s default choice of one per
ten seconds (`-s 10s'). 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, where
decimal points are locale-dependent, e.g. `-s 9.5s' or `-s 9,5s'. Several -S options may be
given; the resulting SEEKTABLE will contain all seekpoints specified (duplicates removed). Note:
`-S #x' and `-S #s' will not work if the encoder cannot determine the input size before starting.
Note: if you use `-S #' with # being >= the number of 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). Use --no-seektable for no SEEKTABLE.

-P #, --padding=#
(Default: 8192 bytes, although 65536 for input above 20 minutes.) Tell the encoder to write a
PADDING metadata block of the given length (in bytes) after the STREAMINFO block. This is useful
for later tagging, where one can write over the PADDING block instead of having to rewrite the
entire file. Note that a block header of 4 bytes will come on top of the length specified.

-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 content if
necessary. This option may appear more than once to add several Vorbis 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.

FORMATOPTIONS
Encoding defaults to FLAC and not OGG. Decoding defaults to WAVE (more specifically WAVE_FORMAT_PCM for
mono/stereo with 8/16 bits, and to WAVE_FORMAT_EXTENSIBLE otherwise), except: will be overridden by
chunks found by --keep-foreign-metadata-if-present or --keep-foreign-metadata

--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.

--force-aiff-format--force-rf64-format--force-wave64-format : For decoding: Override default output format and force output to
AIFF/RF64/WAVE64, respectively. This option is not needed if the output filename (as set by -o) ends
with .aif or .aiff, .rf64 and .w64 respectively. The encoder auto-detects format and ignores this
option.

--force-legacy-wave-format--force-extensible-wave-format : Instruct the decoder to output a WAVE file with WAVE_FORMAT_PCM and
WAVE_FORMAT_EXTENSIBLE respectively, overriding default choice.

--force-aiff-c-none-format--force-aiff-c-sowt-format : Instruct the decoder to output an AIFF-C file with format NONE and sowt
respectively.

--force-raw-format
Force input (when encoding) or output (when decoding) to be treated as raw samples (even if
filename suggests otherwise).

rawformatoptions
When encoding from or decoding to raw PCM, format must be specified.

--sign={signed|unsigned}
Specify the sign of samples.

--endian={big|little}
Specify the byte order for samples

--channels=#
(Input only) specify number of channels. The channels must be interleaved, and in the order of
the FLAC format (see the format specification); the encoder (/decoder) cannot re-order channels.

--bps=#
(Input only) specify bits per sample (per channel: 16 for CDDA.)

--sample-rate=#
(Input only) specify sample rate (in Hz. Only integers supported.)

--input-size=#
(Input only) specify the size of the raw input in bytes. This option is only compulsory when
encoding from stdin and using options that need to know the input size beforehand (like, --skip,
--until, --cuesheet ) The encoder will truncate at the specified size if the input stream is
bigger. If the input stream is smaller, it will complain about an unexpected end-of-file.

ANALYSISOPTIONS--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. gnuplot must be installed separately.

NEGATIVEOPTIONS
The following will negate an option previously given:

--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-seektable--no-silent--no-verify--no-warnings-as-errorsReplayGainapplicationspecification
The option --apply-replaygain-which-is-not-lossless[=<specification>] applies ReplayGain values while
decoding. WARNING:THISISNOTLOSSLESS.DECODEDAUDIOWILLNOTBEIDENTICALTOTHEORIGINALWITHTHISOPTION. This option is useful for example in transcoding media servers, where the client does not support
ReplayGain.

The <specification> is a shorthand notation for describing how to apply ReplayGain. All elements are
optional - defaulting to 0aLn1 - but order is important. 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.

Picturespecification
This described the specification used for the --picture option.
[TYPE]|[MIME-TYPE]|[DESCRIPTION]|[WIDTHxHEIGHTxDEPTH[/COLORS]]|FILETYPE 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 -->

Specificationexamples: “|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. “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.

Apodizationfunctions
To improve LPC analysis, the audio data is windowed. An -A option applies the specified apodization
function(s) instead of the default (which is “tukey(5e-1)”, though different for presets -6 to -8.)
Specifying one more function effectively means, for each subframe, to try another weighting of the data
and see if it happens to result in a smaller encoded subframe. Specifying several functions is
time-expensive, at typically diminishing compression gains.

The subdivide_tukey(N) functions (see below) used in presets -6 to -8 were developed to recycle
calculations for speed, compared to using a number of independent functions. Even then, a high number
like N>4 or 5, will often become less efficient than other options considered expensive, like the slower
-p, though results vary with signal.

Up to 32 functions can be given as comma-separated list and/or individual -A options. Any mis-specified
function is silently ignored. Quoting a function which takes options (and has parentheses) may be
necessary, depending on shell. Currently the following functions are implemented: 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 parameters P, STDDEV and OV, scientific notation is supported, e.g. tukey(5e-1). Otherwise, the
decimal point must agree with the locale, e.g. tukey(0.5) or tukey(0,5) depending on your system.

• For gauss(STDDEV), STDDEV is the standard deviation (0<STDDEV<=5e-1).

• For tukey(P), P (between 0 and 1) specifies the fraction of the window that is cosine-tapered; P=0
corresponds to “rectangle” and P=1 to “hann”.

• partial_tukey(N) and punchout_tukey(N) are largely obsoleted by the more time-effective
subdivide_tukey(N), see next item. They generate N functions each spanning a part of each block.
Optional arguments are an overlap OV (<1, may be negative), for example partial_tukey(2/2e-1); and then
a taper parameter P, for example partial_tukey(2/2e-1/5e-1).

• subdivide_tukey(N) is a more efficient reimplementation of partial_tukey and punchout_tukey taken
together, combining the windows they would generate up to the specified N. Specifying
subdivide_tukey(3) entails a tukey, a partial_tukey(2), a partial_tukey(3) and a punchout_tukey(3);
specifying subdivide_tukey(5) will on top of that add a partial_tukey(4), a punchout_tukey(4), a
partial_tukey(5) and a punchout_tukey(5) - but all with tapering chosen to facilitate the re-use of
computation. Thus the P parameter (defaulting to 5e-1) is applied for the smallest used window: For
example, subdivide_tukey(2/5e-1) results in the same taper as that of tukey(25e-2) and
subdivide_tukey(5) in the same taper as of tukey(1e-1).

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
       | -... ]