flac(1) Free Lossless Audio Codec conversion tool flac(1)

flac - Free Lossless Audio Codec

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 | - ... ]

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

flac supports as input RIFF WAVE, Wave64, RF64, AIFF, FLAC or Ogg FLAC format, or raw interleaved samples. The decoder currently can output to RIFF WAVE, Wave64, RF64, or AIFF format, or raw interleaved samples. flac only supports linear PCM samples (in other words, no A-LAW, uLAW, etc.), and the input must be between 4 and 32 bits per sample.

flac assumes that files ending in “.wav” or that have the RIFF WAVE header present are WAVE files, files ending in “.w64” or have the Wave64 header present are Wave64 files, files ending in “.rf64” or have the RF64 header present are RF64 files, files ending in “.aif” or “.aiff” or have the AIFF header present are AIFF files, files ending in “.flac” or have the FLAC header present are FLAC files and files ending in “.oga” or “.ogg” or have the Ogg FLAC header present are Ogg FLAC files.

Other than this, flac makes no assumptions about file extensions, though the convention is that FLAC files have the extension “.flac” (or “.fla” on ancient “8.3” file systems like FAT-16).

Before going into the full command-line description, a few other things help to sort it out: 1. flac encodes by default, so you must use -d to decode 2. the options -0 .. -8 (or –fast and –best) that control the compression level actually are just synonyms for different groups of specific encoding options (described later) and you can get the same effect by using the same options. When specific options are specified they take priority over the compression level no matter the order 3. flac behaves similarly to gzip in the way it handles input and output files 4. the order in which options are specified is generally not important

Skip to the examples below for examples of some common tasks.

flac will be invoked one of four ways, depending on whether you are encoding, decoding, testing, or analyzing. Encoding is the default invocation, but can be switch to decoding with -d, analysis with -a or testing with -t. Depending on which way is chosen, encoding, decoding, analysis or testing options can be used, see section OPTIONS for details. General options can be used for all.

If only one inputfile is specified, it may be “-” for stdin. When stdin is used as input, flac will write to stdout. Otherwise flac will perform the desired operation on each input file to similarly named output files (meaning for encoding, the extension will be replaced with “.flac”, or appended with “.flac” if the input file has no extension, and for decoding, the extension will be “.wav” for WAVE output and “.raw” for raw output). The original file is not deleted unless –delete-input-file is specified.

If you are encoding/decoding from stdin to a file, you should use the -o option like so:

flac [options] -o outputfile
flac -d [options] -o outputfile

which are better than:

flac [options] > outputfile
flac -d [options] > outputfile

since the former allows flac to seek backwards to write the STREAMINFO or RIFF WAVE header contents when necessary.

Also, you can force output data to go to stdout using -c.

To encode or decode files that start with a dash, use – to signal the end of options, to keep the filenames themselves from being treated as options:

flac -V -- -01-filename.wav

The encoding options affect the compression ratio and encoding speed. The format options are used to tell flac the arrangement of samples if the input file (or output file when decoding) is a raw file. If it is a RIFF WAVE, Wave64, RF64, or AIFF file the format options are not needed since they are read from the file’s header.

In test mode, flac acts just like in decode mode, except no output file is written. Both decode and test modes detect errors in the stream, but they also detect when the MD5 signature of the decoded audio does not match the stored MD5 signature, even when the bitstream is valid.

flac can also re-encode FLAC files. In other words, you can specify a FLAC or Ogg FLAC file as an input to the encoder and it will decoder it and re-encode it according to the options you specify. It will also preserve all the metadata unless you override it with other options (e.g. specifying new tags, seekpoints, cuesheet, padding, etc.).

flac has been tuned so that the default settings yield a good speed vs. compression tradeoff for many kinds of input. However, if you are looking to maximize the compression rate or speed, or want to use the full power of FLAC’s metadata system, see the page titled `About the FLAC Format' on the FLAC website.

Some common encoding tasks using flac:

Encode abc.wav to abc.flac using the default compression setting. abc.wav is not deleted.
Like above, except abc.wav is deleted if there were no errors.
Like above, except abc.wav is deleted if there were no errors or warnings.
Encode abc.wav to abc.flac using the highest compression setting.
Encode abc.wav to abc.flac and internally decode abc.flac to make sure it matches abc.wav.
Encode abc.wav to my.flac.
Encode abc.wav and add some tags at the same time to abc.flac.
Encode all .wav files in the current directory.
Encode abc.aiff to abc.flac.
Encode abc.rf64 to abc.flac.
Encode abc.w64 to abc.flac.
This one’s a little tricky: notice that flac is in encode mode by default (you have to specify -d to decode) so this command actually recompresses abc.flac back to abc.flac. –force is needed to make sure you really want to overwrite abc.flac with a new version. Why would you want to do this? It allows you to recompress an existing FLAC file with (usually) higher compression options or a newer version of FLAC and preserve all the metadata like tags too.

Some common decoding tasks using flac:

Decode abc.flac to abc.wav. abc.flac is not deleted. NOTE: Without -d it means re-encode abc.flac to abc.flac (see above).

flac -d --force-aiff-format abc.flac

flac -d -o abc.aiff abc.flac : Two different ways of decoding abc.flac to abc.aiff (AIFF format). abc.flac is not deleted.

flac -d --force-rf64-format abc.flac

flac -d -o abc.rf64 abc.flac : Two different ways of decoding abc.flac to abc.rf64 (RF64 format). abc.flac is not deleted.

flac -d --force-wave64-format abc.flac

flac -d -o abc.w64 abc.flac : Two different ways of decoding abc.flac to abc.w64 (Wave64 format). abc.flac is not deleted.

Decode abc.flac to abc.wav and don’t abort if errors are found (useful for recovering as much as possible from corrupted files).

A summary of options is included below. For a complete description, see the HTML documentation.

Show the flac version number
Show basic usage and a list of all options
Show detailed explanation of usage and all options
Decode (the default behavior is to encode)
Test a flac encoded file (same as -d except no decoded file is written)
Analyze a FLAC encoded file (same as -d except an analysis file is written)
Write output to stdout
Silent mode (do not write runtime encode/decode statistics to stderr)
Do not print anything of any kind, including warnings or errors. The exit code will be the only way to determine successful completion.
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!
Treat all warnings as errors (which cause flac to terminate with a non-zero exit code).
Force overwriting of output files. By default, flac warns that the output file already exists and continues to the next file.
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.
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).
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.
Output files have their timestamps/permissions set to match those of their inputs (this is default). Use --no-preserve-modtime to make output files have the current time and default permissions.
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.
Like --keep-foreign-metadata, but without throwing an error if foreign metadata cannot be found or restored, instead printing a warning.
Skip over the first number of samples of the input. This works for both encoding and decoding, but not testing. The alternative form mm:ss.ss can be used to specify minutes, seconds, and fractions of a second.
Stop at the given sample number for each input file. This works for both encoding and decoding, but not testing. The given sample number is not included in the decoded output. The alternative form mm:ss.ss can be used to specify minutes, seconds, and fractions of a second. If a `+’ (plus) sign is at the beginning, the --until point is relative to the --skip point. If a `-’ (minus) sign is at the beginning, the --until point is relative to end of the audio.
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'.
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.

Includes the residual signal in the analysis file. This will make the file very big, much larger than even the decoded file.
Generates a gnuplot file for every subframe; each file will contain the residual distribution of the subframe. This will create a lot of files.

Set the beginning and ending cuepoints to decode. The optional first #.# is the track and index point at which decoding will start; the default is the beginning of the stream. The optional second #.# is the track and index point at which decoding will end; the default is the end of the stream. If the cuepoint does not exist, the closest one before it (for the start point) or after it (for the end point) will be used. If those don’t exist, the start of the stream (for the start point) or end of the stream (for the end point) will be used. The cuepoints are merely translated into sample numbers then used as --skip and --until. A CD track can always be cued by, for example, --cue=9.1-10.1 for track 9, even if the CD has no 10th track.
By default flac stops decoding with an error and removes the partially decoded file if it encounters a bitstream error. With -F, errors are still printed but flac will continue decoding to completion. Note that errors may cause the decoded audio to be missing some samples or have silent sections.
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.

Verify a correct encoding by decoding the output in parallel and comparing to the original
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.
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).
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.
Import a picture and store it in a PICTURE metadata block. More than one --picture option can be specified. Either a filename for the picture file or a more complete specification form can be used. The SPECIFICATION is a string whose parts are separated by | (pipe) characters. Some parts may be left empty to invoke default values. FILENAME is just shorthand for “||||FILENAME”. For the format of SPECIFICATION, see the section picture specification.
When encoding to flac, ignore the file size headers in WAV and AIFF files to attempt to work around problems with over-sized or malformed files. WAV and AIFF files both have an unsigned 32 bit numbers in the file header which specifes the length of audio data. Since this number is unsigned 32 bits, that limits the size of a valid file to being just over 4 Gigabytes. Files larger than this are mal-formed, but should be read correctly using this option.
Include a point or points in a SEEKTABLE. Using #, a seek point at that sample number is added. Using X, a placeholder point is added at the end of a the table. Using #x, # evenly spaced seek points will be added, the first being at sample 0. Using #s, a seekpoint will be added every # seconds (# does not have to be a whole number; it can be, for example, 9.5, meaning a seekpoint every 9.5 seconds). You may use many -S options; the resulting SEEKTABLE will be the unique-ified union of all such values. With no -S options, flac defaults to `-S 10s'. Use --no-seektable for no SEEKTABLE. Note: `-S #x' and `-S #s' will not work if the encoder can’t determine the input size before starting. Note: if you use `-S #' and # is >= samples in the input, there will be either no seek point entered (if the input size is determinable before encoding starts) or a placeholder point (if input size is not determinable).
Tell the encoder to write a PADDING metadata block of the given length (in bytes) after the STREAMINFO block. This is useful if you plan to tag the file later with an APPLICATION block; instead of having to rewrite the entire file later just to insert your block, you can write directly over the PADDING block. Note that the total length of the PADDING block will be 4 bytes longer than the length given because of the 4 metadata block header bytes. You can force no PADDING block at all to be written with --no-padding. The encoder writes a PADDING block of 8192 bytes by default (or 65536 bytes if the input audio stream is more that 20 minutes long).
Add a FLAC tag. The comment must adhere to the Vorbis comment spec; i.e. the FIELD must contain only legal characters, terminated by an `equals' sign. Make sure to quote the comment if necessary. This option may appear more than once to add several comments. NOTE: all tags will be added to all encoded files.
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.
Specify the blocksize in samples. The default is 1152 for -l 0, else 4096. For subset streams this must be <= 4608 if the samplerate <= 48kHz, for subset streams with higher samplerates it must be <= 16384.
Try mid-side coding for each frame (stereo input only)
Adaptive mid-side coding for all frames (stereo input only)
-0..-8, --compression-level-0..--compression-level-8
Fastest compression..highest compression (default is -5). These are synonyms for other options:
-0, --compression-level-0
Synonymous with -l 0 -b 1152 -r 3 --no-mid-side
-1, --compression-level-1
Synonymous with -l 0 -b 1152 -M -r 3
-2, --compression-level-2
Synonymous with -l 0 -b 1152 -m -r 3
-3, --compression-level-3
Synonymous with -l 6 -b 4096 -r 4 --no-mid-side
-4, --compression-level-4
Synonymous with -l 8 -b 4096 -M -r 4
-5, --compression-level-5
Synonymous with -l 8 -b 4096 -m -r 5
-6, --compression-level-6
Synonymous with -l 8 -b 4096 -m -r 6 -A subdivide_tukey(2)
-7, --compression-level-7
Synonymous with -l 12 -b 4096 -m -r 6 -A subdivide_tukey(2)
-8, --compression-level-8
Synonymous with -l 12 -b 4096 -m -r 6 -A subdivide_tukey(3)
Fastest compression. Currently synonymous with -0.
Highest compression. Currently synonymous with -8.
Do exhaustive model search (expensive!)
Window audio data with given the apodization function. See section Apodization functions for details.
Specifies the maximum LPC order. This number must be <= 32. For subset streams, it must be <=12 if the sample rate is <=48kHz. If 0, the encoder will not attempt generic linear prediction, and use only fixed predictors. Using fixed predictors is faster but usually results in files being 5-10% larger.
Do exhaustive search of LP coefficient quantization (expensive!). Overrides -q; does nothing if using -l 0
Precision of the quantized linear-predictor coefficients, 0 => let encoder decide (min is 5, default is 0)
Set the [min,]max residual partition order (0..15). min defaults to 0 if unspecified. Default is -r 5.

Set the byte order for samples
Set number of channels.
Set bits per sample.
Set sample rate (in Hz).
Set the sign of samples.
Specify the size of the raw input in bytes. If you are encoding raw samples from stdin, you must set this option in order to be able to use --skip, --until, --cuesheet, or other options that need to know the size of the input beforehand. If the size given is greater than what is found in the input stream, the encoder will complain about an unexpected end-of-file. If the size given is less, samples will be truncated.
Force input (when encoding) or output (when decoding) to be treated as raw samples (even if filename ends in .wav).

--force-aiff-format

--force-rf64-format

--force-wave64-format : Force the decoder to output AIFF/RF64/WAVE64 format respectively. This option is not needed if the output filename (as set by -o) ends with .aif or .aiff, .rf64 and .w64 respectively. Also, this option has no effect when encoding since input is auto-detected. When none of these options nor –keep-foreign-metadata are given and no output filename is set, the output format is WAV by default.

--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. If none of these options nor –keep-foreign-metadata are given, FLAC outputs WAVE_FORMAT_PCM for mono or stereo with a bit depth of 8 or 16 bits, and WAVE_FORMAT_EXTENSIBLE for all other audio formats.

--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.

--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

These flags can be used to invert the sense of the corresponding normal option.

The option --apply-replaygain-which-is-not-lossless[=<specification>] applies ReplayGain values while decoding. WARNING: THIS IS NOT LOSSLESS. DECODED AUDIO WILL NOT BE IDENTICAL TO THE ORIGINAL WITH THIS OPTION.** This option is useful for example in transcoding media servers, where the client does not support ReplayGain.

The equals sign and <specification> is optional. If omitted, the default specification is 0aLn1.

The <specification> is a shorthand notation for describing how to apply ReplayGain. All components are optional but order is important. `[]' means `optional'. `|' means `or'. `{}' means required. The format is:

[<preamp>][a|t][l|L][n{0|1|2|3}]

In which the following parameters are used:

•
preamp: A floating point number in dB. This is added to the existing gain value.
•
a|t: Specify `a' to use the album gain, or `t' to use the track gain. If tags for the preferred kind (album/track) do not exist but tags for the other (track/album) do, those will be used instead.
•
l|L: Specify `l' to peak-limit the output, so that the ReplayGain peak value is full-scale. Specify `L' to use a 6dB hard limiter that kicks in when the signal approaches full-scale.
•
n{0|1|2|3}: Specify the amount of noise shaping. ReplayGain synthesis happens in floating point; the result is dithered before converting back to integer. This quantization adds noise. Noise shaping tries to move the noise where you won’t hear it as much. 0 means no noise shaping, 1 means `low', 2 means `medium', 3 means `high'.

For example, the default of 0aLn1 means 0dB preamp, use album gain, 6dB hard limit, low noise shaping. --apply-replaygain-which-is-not-lossless=3 means 3dB preamp, use album gain, no limiting, no noise shaping.

flac uses the ReplayGain tags for the calculation. If a stream does not have the required tags or they can’t be parsed, decoding will continue with a warning, and no ReplayGain is applied to that stream.

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

TYPE is optional; it is a number from one of:

0.
Other
1.
32x32 pixels `file icon' (PNG only)
2.
Other file icon
3.
Cover (front)
4.
Cover (back)
5.
Leaflet page
6.
Media (e.g. label side of CD)
7.
Lead artist/lead performer/soloist
8.
Artist/performer
9.
Conductor
10.
Band/Orchestra
11.
Composer
12.
Lyricist/text writer
13.
Recording Location
14.
During recording
15.
During performance
16.
Movie/video screen capture
17.
A bright coloured fish
18.
Illustration
19.
Band/artist logotype
20.
Publisher/Studio logotype

The default is 3 (front cover). There may only be one picture each of type 1 and 2 in a file.

MIME-TYPE is optional; if left blank, it will be detected from the file. For best compatibility with players, use pictures with MIME type image/jpeg or image/png. The MIME type can also be --> to mean that FILE is actually a URL to an image, though this use is discouraged.

DESCRIPTION is optional; the default is an empty string.

The next part specifies the resolution and color information. If the MIME-TYPE is image/jpeg, image/png, or image/gif, you can usually leave this empty and they can be detected from the file. Otherwise, you must specify the width in pixels, height in pixels, and color depth in bits-per-pixel. If the image has indexed colors you should also specify the number of colors used. When manually specified, it is not checked against the file for accuracy.

FILE is the path to the picture file to be imported, or the URL if MIME type is -->

For example, “|image/jpeg|||../cover.jpg” will embed the JPEG file at ../cover.jpg, defaulting to type 3 (front cover) and an empty description. The resolution and color info will be retrieved from the file itself.

The specification “4|-->|CD|320x300x24/173|http://blah.blah/backcover.tiff” will embed the given URL, with type 4 (back cover), description “CD”, and a manually specified resolution of 320x300, 24 bits-per-pixel, and 173 colors. The file at the URL will not be fetched; the URL itself is stored in the PICTURE metadata block.

To improve LPC analysis, audio data is windowed . The window can be selected with one or more -A options. Possible functions are: bartlett, bartlett_hann, blackman, blackman_harris_4term_92db, connes, flattop, gauss(STDDEV), hamming, hann, kaiser_bessel, nuttall, rectangle, triangle, tukey(P), partial_tukey(n[/ov[/P]]), punchout_tukey(n[/ov[/P]]), subdivide_tukey(n[/P]) welch.

•
For gauss(STDDEV), STDDEV is the standard deviation (0<STDDEV<=0.5).
•
For tukey(P), P specifies the fraction of the window that is tapered (0<=P<=1; P=0 corresponds to “rectangle” and P=1 corresponds to “hann”).
•
For partial_tukey(n) and punchout_tukey(n), n apodization functions are added that span different parts of each block. Values of 2 to 6 seem to yield sane results. If necessary, an overlap can be specified, as can be the taper parameter, for example partial_tukey(2/0.2) or partial_tukey(2/0.2/0.5). ov should be smaller than 1 and can be negative. The use of this is that different parts of a block are ignored as the might contain transients which are hard to predict anyway. The encoder will try each different added apodization (each covering a different part of the block) to see which resulting predictor results in the smallest representation.
•
subdivide_tukey(n) is a more efficient reimplementation of partial_tukey and punchout_tukey taken together, recycling as much data as possible. It combines all possible non-redundant partial_tukey(n) and punchout_tukey(n) up to the n specified. Specifying subdivide_tukey(3) is equivalent to specifying tukey, partial_tukey(2), partial_tukey(3) and punchout_tukey(3), specifying subdivide_tukey(5) equivalently adds partial_tukey(4), punchout_tukey(4), partial_tukey(5) and punchout_tukey(5). To be able to reuse data as much as possible, the tukey taper is taken equal for all windows, and the P specified is applied for the smallest used window. In other words, subdivide_tukey(2/0.5) results in a taper equal to that of tukey(0.25) and subdivide_tukey(5) in a taper equal to that of tukey(0.1). The default P for subdivide_tukey when none is specified is 0.5.

Note that P, STDDEV and ov are locale specific, so a comma as decimal separator might be required instead of a dot. Use scientific notation for a locale-independent specification, for example tukey(5e-1) instead of tukey(0.5) or tukey(0,5).

More than one -A option (up to 32) may be used. Any function that is specified erroneously is silently dropped. The encoder chooses suitable defaults in the absence of any -A options; any -A option specified replaces the default(s).

When more than one function is specified, then for every subframe the encoder will try each of them separately and choose the window that results in the smallest compressed subframe. Multiple functions can greatly increase the encoding time.

metaflac(1)

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 1.4.3