flac(1) Free Lossless Audio Codec conversion tool flac(1)
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. 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.
EXAMPLES
Some typical encoding and decoding tasks using flac:
Encoding examples
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.
Decoding examples
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 –decode or its shortform -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.
OPTIONS
A summary of options is included below. Several of the options can be
negated, see the Negative options section below.
GENERAL OPTIONS
-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.
-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).
--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).
DECODING OPTIONS
-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: 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
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.
-A FUNCTION(S), --apodization=FUNCTION(S)
Window audio data with given apodization function. More can be
given, comma-separated. See section Apodization functions 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 Picture
specification 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.
FORMAT OPTIONS
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).
raw format options
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.
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. gnuplot must be installed separately.
NEGATIVE OPTIONS
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-errors
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 <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.
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 -->
Specification examples: “|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.
Apodization functions
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).
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.
Version 1.5.0 flac(1)
flac \- Free Lossless Audio Codec