furnace/extern/flac/man/flac.html
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<header id="title-block-header">
<h1 class="title">flac(1) Version 1.5.0 | Free Lossless Audio Codec
conversion tool</h1>
</header>
<h1 id="name">NAME</h1>
<p>flac - Free Lossless Audio Codec</p>
<h1 id="synopsis">SYNOPSIS</h1>
<p><strong>flac</strong> [ <em>OPTIONS</em> ] [ <em>infile.wav</em> |
<em>infile.rf64</em> | <em>infile.aiff</em> | <em>infile.raw</em> |
<em>infile.flac</em> | <em>infile.oga</em> | <em>infile.ogg</em> |
<strong>-</strong> <em></em> ]</p>
<p><strong>flac</strong> [ <strong>-d</strong> |
<strong>--decode</strong> | <strong>-t</strong> |
<strong>--test</strong> | <strong>-a</strong> |
<strong>--analyze</strong> ] [ <em>OPTIONS</em> ] [ <em>infile.flac</em>
| <em>infile.oga</em> | <em>infile.ogg</em> | <strong>-</strong>
<em></em> ]</p>
<h1 id="description">DESCRIPTION</h1>
<p><strong>flac</strong> is a command-line tool for encoding, decoding,
testing and analyzing FLAC streams.</p>
<h1 id="general-usage">GENERAL USAGE</h1>
<p><strong>flac</strong> 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.</p>
<p>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.</p>
<p>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).</p>
<p>Before going into the full command-line description, a few other
things help to sort it out:</p>
<ol type="1">
<li>flac encodes by default, so you must use -d to decode</li>
<li>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).<br />
</li>
<li>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.</li>
<li>flac behaves similarly to gzip in the way it handles input and
output files</li>
</ol>
<p>Skip to the EXAMPLES section below for examples of some typical
tasks.</p>
<p>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 <strong>-d</strong>,
analysis with <strong>-a</strong> or testing with <strong>-t</strong>.
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.</p>
<p>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.</p>
<p>If you are encoding/decoding from stdin to a file, you should use the
-o option like so:</p>
<pre><code>flac [options] -o outputfile
flac -d [options] -o outputfile</code></pre>
<p>which are better than:</p>
<pre><code>flac [options] &gt; outputfile
flac -d [options] &gt; outputfile</code></pre>
<p>since the former allows flac to seek backwards to write the
STREAMINFO or RIFF WAVE header contents when necessary.</p>
<p>Also, you can force output data to go to stdout using -c.</p>
<p>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:</p>
<pre><code>flac -V -- -01-filename.wav</code></pre>
<p>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 files header.</p>
<p>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.</p>
<p>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.).</p>
<p>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 FLACs metadata system, see the page titled About the
FLAC Format on the FLAC website.</p>
<h1 id="examples">EXAMPLES</h1>
<p>Some typical encoding and decoding tasks using flac:</p>
<h2 id="encoding-examples">Encoding examples</h2>
<dl>
<dt><code>flac abc.wav</code></dt>
<dd>
Encode abc.wav to abc.flac using the default compression setting.
abc.wav is not deleted.
</dd>
<dt><code>flac --delete-input-file abc.wav</code></dt>
<dd>
Like above, except abc.wav is deleted if there were no errors.
</dd>
<dt><code>flac --delete-input-file -w abc.wav</code></dt>
<dd>
Like above, except abc.wav is deleted if there were no errors and no
warnings.
</dd>
<dt><code>flac --best abc.wav</code> or
<code>flac -8 abc.wav</code></dt>
<dd>
Encode abc.wav to abc.flac using the highest compression preset.
</dd>
<dt><code>flac --verify abc.wav</code> or
<code>flac -V abc.wav</code></dt>
<dd>
Encode abc.wav to abc.flac and internally decode abc.flac to make sure
it matches abc.wav.
</dd>
<dt><code>flac -o my.flac abc.wav</code></dt>
<dd>
Encode abc.wav to my.flac.
</dd>
<dt><code>flac abc.aiff foo.rf64 bar.w64</code></dt>
<dd>
Encode abc.aiff to abc.flac, foo.rf64 to foo.flac and bar.w64 to
bar.flac
</dd>
<dt><code>flac *.wav *.aif?</code></dt>
<dd>
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.
</dd>
<dt><code>flac abc.flac --force</code> or
<code>flac abc.flac -f</code></dt>
<dd>
Recompresses, keeping metadata like tags. The syntax is a little tricky:
this is an <em>encoding</em> 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.
</dd>
<dt><code>flac --tag-from-file=&quot;ALBUM=albumtitle.txt&quot; -T &quot;ARTIST=Queen&quot; *.wav</code></dt>
<dd>
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.
</dd>
<dt><code>flac --keep-foreign-metadata-if-present abc.wav</code></dt>
<dd>
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.
</dd>
<dt><code>flac -Vj2 -m3fo Track07.flac -- -7.wav</code></dt>
<dd>
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 <code>--</code> (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.
</dd>
</dl>
<h2 id="decoding-examples">Decoding examples</h2>
<dl>
<dt><code>flac --decode abc.flac</code> or
<code>flac -d abc.flac</code></dt>
<dd>
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 <em>without decode or its shortform -d</em>, would mean to
re-encode abc.flac to abc.flac (see above), and that command would err
out because abc.flac already exists.
</dd>
<dt><code>flac -d --force-aiff-format abc.flac</code> or
<code>flac -d -o abc.aiff abc.flac</code></dt>
<dd>
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.
</dd>
<dt><code>flac -d --keep-foreign-metadata-if-present abc.flac</code></dt>
<dd>
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.
</dd>
<dt><code>flac -d -F abc.flac</code></dt>
<dd>
Decode abc.flac to abc.wav and dont 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.
</dd>
</dl>
<h1 id="options">OPTIONS</h1>
<p>A summary of options is included below. Several of the options can be
negated, see the <strong>Negative options</strong> section below.</p>
<h2 id="general-options">GENERAL OPTIONS</h2>
<dl>
<dt><strong>-v</strong>, <strong>--version</strong></dt>
<dd>
Show the flac version number, and quit.
</dd>
<dt><strong>-h</strong>, <strong>--help</strong></dt>
<dd>
Show basic usage and a list of all options, and quit.
</dd>
<dt><strong>-d</strong>, <strong>--decode</strong></dt>
<dd>
Decode (the default behavior is to encode)
</dd>
<dt><strong>-t</strong>, <strong>--test</strong></dt>
<dd>
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.
</dd>
<dt><strong>-a</strong>, <strong>--analyze</strong></dt>
<dd>
Analyze a FLAC encoded file. This works the same as -d except the output
is an analysis file, not a decoded file.
</dd>
<dt><strong>-c</strong>, <strong>--stdout</strong></dt>
<dd>
Write output to stdout
</dd>
<dt><strong>-f</strong>, <strong>--force</strong></dt>
<dd>
Force overwriting of output files. By default, flac warns that the
output file already exists and continues to the next file.
</dd>
<dt><strong>--delete-input-file</strong></dt>
<dd>
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.
</dd>
<dt><strong>-o</strong> <em>FILENAME</em>,
<strong>--output-name</strong>=<em>FILENAME</em></dt>
<dd>
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.
</dd>
<dt><strong>--output-prefix</strong>=<em>STRING</em></dt>
<dd>
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).
</dd>
<dt><strong>--preserve-modtime</strong></dt>
<dd>
(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.
</dd>
<dt><strong>--keep-foreign-metadata</strong></dt>
<dd>
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.
</dd>
<dt><strong>--keep-foreign-metadata-if-present</strong></dt>
<dd>
Like --keep-foreign-metadata, but without throwing an error if foreign
metadata cannot be found or restored. Instead, prints a warning.
</dd>
<dt><strong>--skip</strong>={#|<em>MM:SS</em>}</dt>
<dd>
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.
</dd>
<dt><strong>--until</strong>={#|[+|]<em>MM:SS</em>}</dt>
<dd>
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.
</dd>
<dt><strong>--no-utf8-convert</strong></dt>
<dd>
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!
</dd>
<dt><strong>-s</strong>, <strong>--silent</strong></dt>
<dd>
Silent mode (do not write runtime encode/decode statistics to stderr)
</dd>
<dt><strong>--totally-silent</strong></dt>
<dd>
Do not print anything of any kind, including warnings or errors. The
exit code will be the only way to determine successful completion.
</dd>
<dt><strong>-w</strong>, <strong>--warnings-as-errors</strong></dt>
<dd>
Treat all warnings as errors (which cause flac to terminate with a
non-zero exit code).
</dd>
</dl>
<h2 id="decoding-options">DECODING OPTIONS</h2>
<dl>
<dt><strong>-F</strong>, <strong>--decode-through-errors</strong></dt>
<dd>
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.
</dd>
<dt><strong>--cue</strong>=[#.#][-[#.#]]</dt>
<dd>
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 dont 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.
</dd>
<dt><strong>decode-chained-stream</strong></dt>
<dd>
Decode all links in a chained Ogg stream, not just the first one.
</dd>
<dt><strong>--apply-replaygain-which-is-not-lossless</strong>[=<em>SPECIFICATION</em>]</dt>
<dd>
Applies ReplayGain values while decoding. <strong>WARNING: THIS IS NOT
LOSSLESS. DECODED AUDIO WILL NOT BE IDENTICAL TO THE ORIGINAL WITH THIS
OPTION.</strong> 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 <strong>ReplayGain application
specification</strong>.
</dd>
</dl>
<h2 id="encoding-options">ENCODING OPTIONS</h2>
<p>Encoding will default to -5, -A “tukey(5e-1)” and one CPU thread.</p>
<dl>
<dt><strong>-V</strong>, <strong>--verify</strong></dt>
<dd>
Verify a correct encoding by decoding the output in parallel and
comparing to the original.
</dd>
<dt><strong>-0</strong>, <strong>--compression-level-0</strong>,
<strong>--fast</strong></dt>
<dd>
Fastest compression preset. Currently synonymous with
<code>-l 0 -b 1152 -r 3 --no-mid-side</code>
</dd>
<dt><strong>-1</strong>, <strong>--compression-level-1</strong></dt>
<dd>
Currently synonymous with <code>-l 0 -b 1152 -M -r 3</code>
</dd>
<dt><strong>-2</strong>, <strong>--compression-level-2</strong></dt>
<dd>
Currently synonymous with <code>-l 0 -b 1152 -m -r 3</code>
</dd>
<dt><strong>-3</strong>, <strong>--compression-level-3</strong></dt>
<dd>
Currently synonymous with <code>-l 6 -b 4096 -r 4 --no-mid-side</code>
</dd>
<dt><strong>-4</strong>, <strong>--compression-level-4</strong></dt>
<dd>
Currently synonymous with <code>-l 8 -b 4096 -M -r 4</code>
</dd>
<dt><strong>-5</strong>, <strong>--compression-level-5</strong></dt>
<dd>
Currently synonymous with <code>-l 8 -b 4096 -m -r 5</code>
</dd>
<dt><strong>-6</strong>, <strong>--compression-level-6</strong></dt>
<dd>
Currently synonymous with
<code>-l 8 -b 4096 -m -r 6 -A &quot;subdivide_tukey(2)&quot;</code>
</dd>
<dt><strong>-7</strong>, <strong>--compression-level-7</strong></dt>
<dd>
Currently synonymous with
<code>-l 12 -b 4096 -m -r 6 -A &quot;subdivide_tukey(2)&quot;</code>
</dd>
<dt><strong>-8</strong>, <strong>--compression-level-8</strong>,
<strong>--best</strong></dt>
<dd>
Currently synonymous with
<code>-l 12 -b 4096 -m -r 6 -A &quot;subdivide_tukey(3)&quot;</code>
</dd>
<dt><strong>-l</strong> #, <strong>--max-lpc-order</strong>=#</dt>
<dd>
Specifies the maximum LPC order. This number must be &lt;= 32. For
subset streams, it must be &lt;=12 if the sample rate is &lt;=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.
</dd>
<dt><strong>-b</strong> #, <strong>--blocksize</strong>=#</dt>
<dd>
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 &lt;= 4608 if the samplerate is &lt;= 48kHz,
for subset streams with higher samplerates it must be &lt;= 16384.
</dd>
<dt><strong>-m</strong>, <strong>--mid-side</strong></dt>
<dd>
Try mid-side coding for each frame (stereo only, otherwise ignored).
</dd>
<dt><strong>-M</strong>, <strong>--adaptive-mid-side</strong></dt>
<dd>
Adaptive mid-side coding for all frames (stereo only, otherwise
ignored).
</dd>
<dt><strong>-r</strong> [#,]#,
<strong>--rice-partition-order</strong>=[#,]#</dt>
<dd>
Set the [min,]max residual partition order (0..15). For subset streams,
max must be &lt;=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.
</dd>
<dt><strong>-A</strong> <em>FUNCTION(S)</em>,
<strong>--apodization</strong>=<em>FUNCTION(S)</em></dt>
<dd>
Window audio data with given apodization function. More can be given,
comma-separated. See section <strong>Apodization functions</strong> for
details.
</dd>
<dt><strong>-e</strong>, <strong>--exhaustive-model-search</strong></dt>
<dd>
Do exhaustive model search (expensive!).
</dd>
<dt><strong>-q</strong> #, <strong>--qlp-coeff-precision</strong>=#</dt>
<dd>
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.
</dd>
<dt><strong>-p</strong>,
<strong>--qlp-coeff-precision-search</strong></dt>
<dd>
Do exhaustive search of LP coefficient quantization (expensive!).
Overrides -q; does nothing if using -l 0.
</dd>
<dt><strong>--lax</strong></dt>
<dd>
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.
</dd>
<dt><strong>--limit-min-bitrate</strong></dt>
<dd>
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.
</dd>
<dt><strong>-j</strong> #, <strong>--threads</strong>=#</dt>
<dd>
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.
</dd>
<dt><strong>--ignore-chunk-sizes</strong></dt>
<dd>
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.
</dd>
<dt><strong>--replay-gain</strong></dt>
<dd>
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).
</dd>
<dt><strong>--cuesheet</strong>=<em>FILENAME</em></dt>
<dd>
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.
</dd>
<dt><strong>--picture</strong>={<em>FILENAME|SPECIFICATION</em>}</dt>
<dd>
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
<em>SPECIFICATION</em> is a string whose parts are separated by | (pipe)
characters. Some parts may be left empty to invoke default values.
Specifying only <em>FILENAME</em> is just shorthand for “||||FILENAME”.
See the section <strong>Picture specification</strong> for
<em>SPECIFICATION</em> format.
</dd>
<dt><strong>-S</strong> {#|X|#x|#s},
<strong>--seekpoint</strong>={#|X|#x|#s}</dt>
<dd>
Specifies point(s) to include in SEEKTABLE, to override the encoders
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
&gt;= 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.
</dd>
<dt><strong>-P</strong> #, <strong>--padding</strong>=#</dt>
<dd>
(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.
</dd>
<dt><strong>-T</strong><em>FIELD=VALUE</em><strong>,
--tag</strong>=“<em>FIELD=VALUE</em></dt>
<dd>
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.
</dd>
<dt><strong>--tag-from-file</strong>=“<em>FIELD=FILENAME</em></dt>
<dd>
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.
</dd>
</dl>
<h2 id="format-options">FORMAT OPTIONS</h2>
<p>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</p>
<dl>
<dt><strong>--ogg</strong></dt>
<dd>
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.
</dd>
<dt><strong>--serial-number</strong>=#</dt>
<dd>
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.
</dd>
</dl>
<p><strong>--force-aiff-format</strong><br />
<strong>--force-rf64-format</strong><br />
<strong>--force-wave64-format</strong> : 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
<em>.aif</em> or <em>.aiff</em>, <em>.rf64</em> and <em>.w64</em>
respectively. The encoder auto-detects format and ignores this
option.</p>
<p><strong>--force-legacy-wave-format</strong><br />
<strong>--force-extensible-wave-format</strong> : Instruct the decoder
to output a WAVE file with WAVE_FORMAT_PCM and WAVE_FORMAT_EXTENSIBLE
respectively, overriding default choice.</p>
<p><strong>--force-aiff-c-none-format</strong><br />
<strong>--force-aiff-c-sowt-format</strong> : Instruct the decoder to
output an AIFF-C file with format NONE and sowt respectively.</p>
<dl>
<dt><strong>--force-raw-format</strong></dt>
<dd>
Force input (when encoding) or output (when decoding) to be treated as
raw samples (even if filename suggests otherwise).
</dd>
</dl>
<h3 id="raw-format-options">raw format options</h3>
<p>When encoding from or decoding to raw PCM, format must be
specified.</p>
<dl>
<dt><strong>--sign</strong>={signed|unsigned}</dt>
<dd>
Specify the sign of samples.
</dd>
<dt><strong>--endian</strong>={big|little}</dt>
<dd>
Specify the byte order for samples
</dd>
<dt><strong>--channels</strong>=#</dt>
<dd>
(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.
</dd>
<dt><strong>--bps</strong>=#</dt>
<dd>
(Input only) specify bits per sample (per channel: 16 for CDDA.)
</dd>
<dt><strong>--sample-rate</strong>=#</dt>
<dd>
(Input only) specify sample rate (in Hz. Only integers supported.)
</dd>
<dt><strong>--input-size</strong>=#</dt>
<dd>
(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.
</dd>
</dl>
<h2 id="analysis-options">ANALYSIS OPTIONS</h2>
<dl>
<dt><strong>--residual-text</strong></dt>
<dd>
Includes the residual signal in the analysis file. This will make the
file very big, much larger than even the decoded file.
</dd>
<dt><strong>--residual-gnuplot</strong></dt>
<dd>
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.
</dd>
</dl>
<h2 id="negative-options">NEGATIVE OPTIONS</h2>
<p>The following will negate an option previously given:</p>
<p><strong>--no-adaptive-mid-side</strong><br />
<strong>--no-cued-seekpoints</strong><br />
<strong>--no-decode-through-errors</strong><br />
<strong>--no-delete-input-file</strong><br />
<strong>--no-preserve-modtime</strong><br />
<strong>--no-keep-foreign-metadata</strong><br />
<strong>--no-exhaustive-model-search</strong><br />
<strong>--no-force</strong><br />
<strong>--no-lax</strong><br />
<strong>--no-mid-side</strong><br />
<strong>--no-ogg</strong><br />
<strong>--no-padding</strong><br />
<strong>--no-qlp-coeff-prec-search</strong><br />
<strong>--no-replay-gain</strong><br />
<strong>--no-residual-gnuplot</strong><br />
<strong>--no-residual-text</strong><br />
<strong>--no-seektable</strong><br />
<strong>--no-silent</strong><br />
<strong>--no-verify</strong><br />
<strong>--no-warnings-as-errors</strong></p>
<h2 id="replaygain-application-specification">ReplayGain application
specification</h2>
<p>The option
--apply-replaygain-which-is-not-lossless[=&lt;specification&gt;] applies
ReplayGain values while decoding. <strong>WARNING: THIS IS NOT LOSSLESS.
DECODED AUDIO WILL NOT BE IDENTICAL TO THE ORIGINAL WITH THIS
OPTION.</strong> This option is useful for example in transcoding media
servers, where the client does not support ReplayGain.</p>
<p>The &lt;specification&gt; is a shorthand notation for describing how
to apply ReplayGain. All elements are optional - defaulting to 0aLn1 -
but order is important. The format is:</p>
<p>[&lt;preamp&gt;][a|t][l|L][n{0|1|2|3}]</p>
<p>In which the following parameters are used:</p>
<ul>
<li><p><strong>preamp</strong>: A floating point number in dB. This is
added to the existing gain value.</p></li>
<li><p><strong>a|t</strong>: 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.</p></li>
<li><p><strong>l|L</strong>: 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.</p></li>
<li><p><strong>n{0|1|2|3}</strong>: 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 wont hear it as much. 0 means
no noise shaping, 1 means low, 2 means medium, 3 means
high.</p></li>
</ul>
<p>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.</p>
<p>flac uses the ReplayGain tags for the calculation. If a stream does
not have the required tags or they cant be parsed, decoding will
continue with a warning, and no ReplayGain is applied to that
stream.</p>
<h2 id="picture-specification">Picture specification</h2>
<p>This described the specification used for the
<strong>--picture</strong> option.
[<em>TYPE</em>]|[<em>MIME-TYPE</em>]|[<em>DESCRIPTION</em>]|[<em>WIDTHxHEIGHTxDEPTH</em>[/<em>COLORS</em>]]|<em>FILE</em></p>
<p><em>TYPE</em> is optional; it is a number from one of:</p>
<ol start="0" type="1">
<li>Other</li>
<li>32x32 pixels file icon (PNG only)</li>
<li>Other file icon</li>
<li>Cover (front)</li>
<li>Cover (back)</li>
<li>Leaflet page</li>
<li>Media (e.g. label side of CD)</li>
<li>Lead artist/lead performer/soloist</li>
<li>Artist/performer</li>
<li>Conductor</li>
<li>Band/Orchestra</li>
<li>Composer</li>
<li>Lyricist/text writer</li>
<li>Recording Location</li>
<li>During recording</li>
<li>During performance</li>
<li>Movie/video screen capture</li>
<li>A bright coloured fish</li>
<li>Illustration</li>
<li>Band/artist logotype</li>
<li>Publisher/Studio logotype</li>
</ol>
<p>The default is 3 (front cover). There may only be one picture each of
type 1 and 2 in a file.</p>
<p><em>MIME-TYPE</em> 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 --&gt; to
mean that FILE is actually a URL to an image, though this use is
discouraged.</p>
<p><em>DESCRIPTION</em> is optional; the default is an empty string.</p>
<p>The next part specifies the resolution and color information. If the
<em>MIME-TYPE</em> 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.</p>
<p><em>FILE</em> is the path to the picture file to be imported, or the
URL if MIME type is --&gt;</p>
<p><strong>Specification examples:</strong> “|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|--&gt;|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.</p>
<h2 id="apodization-functions">Apodization functions</h2>
<p>To improve LPC analysis, the audio data is windowed. An
<strong>-A</strong> 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.</p>
<p>The subdivide_tukey(<em>N</em>) 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
<em>N</em>&gt;4 or 5, will often become less efficient than other
options considered expensive, like the slower -p, though results vary
with signal.</p>
<p>Up to 32 functions can be given as comma-separated list and/or
individual <strong>-A</strong> 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(<em>STDDEV</em>),
hamming, hann, kaiser_bessel, nuttall, rectangle, triangle,
tukey(<em>P</em>), partial_tukey(<em>N</em>[/<em>OV</em>[/<em>P</em>]]),
punchout_tukey(<em>N</em>[/<em>OV</em>[/<em>P</em>]]),
subdivide_tukey(<em>N</em>[/<em>P</em>]), welch.</p>
<p>For parameters <em>P</em>, <em>STDDEV</em> and <em>OV</em>,
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.</p>
<ul>
<li><p>For gauss(<em>STDDEV</em>), <em>STDDEV</em> is the standard
deviation (0&lt;<em>STDDEV</em>&lt;=5e-1).</p></li>
<li><p>For tukey(<em>P</em>), <em>P</em> (between 0 and 1) specifies the
fraction of the window that is cosine-tapered; <em>P</em>=0 corresponds
to “rectangle” and <em>P</em>=1 to “hann”.</p></li>
<li><p>partial_tukey(<em>N</em>) and punchout_tukey(<em>N</em>) are
largely obsoleted by the more time-effective
subdivide_tukey(<em>N</em>), see next item. They generate <em>N</em>
functions each spanning a part of each block. Optional arguments are an
overlap <em>OV</em> (&lt;1, may be negative), for example
partial_tukey(2/2e-1); and then a taper parameter <em>P</em>, for
example partial_tukey(2/2e-1/5e-1).</p></li>
<li><p>subdivide_tukey(<em>N</em>) is a more efficient reimplementation
of partial_tukey and punchout_tukey taken together, combining the
windows they would generate up to the specified <em>N</em>. 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 <em>P</em> 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).</p></li>
</ul>
<h1 id="see-also">SEE ALSO</h1>
<p><strong>metaflac(1)</strong></p>
<h1 id="author">AUTHOR</h1>
<p>This manual page was initially written by Matt Zimmerman
&lt;mdz@debian.org&gt; for the Debian GNU/Linux system (but may be used
by others). It has been kept up-to-date by the Xiph.org Foundation.</p>
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