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fdkaac - command line frontend for libfdk-aac encoder


**fdkaac** [OPTIONS] [FILE]


**fdkaac** reads linear PCM audio in either WAV, raw PCM, or CAF format,
and encodes it into either M4A / AAC file.

If the input file is "-", data is read from stdin. Likewise, if the
output file is "-", data is written to stdout if one of streamable AAC
transport formats are selected by **-f**.

When CAF input and M4A output is used, tags in CAF file are copied into
the resulting M4A.


-h, --help
:   Show command help

-o \<FILE\>
:   Output filename.

-p, --profile \<n\>
:   Target profile (MPEG4 audio object type, AOT)

    :   MPEG-4 AAC LC (default)

    :   MPEG-4 HE-AAC (SBR)

    :   MPEG-4 HE-AAC v2 (SBR+PS)

    :   MPEG-4 AAC LD

    :   MPEG-4 AAC ELD

-b, --bitrate \<n\>
:   Target bitrate (for CBR)

-m, --bitrate-mode \<n\>
:   Bitrate configuration mode. Available VBR quality value depends on
    other parameters such as profile, sample rate, or number of

    :   CBR (default)

    :   VBR (higher value -\> higher bitrate)

-w, --bandwidth \<n\>
:   Frequency bandwidth (lowpass cut-off frequency) in Hz. Available on
    AAC LC only.

-a, --afterburner \<n\>
:   Configure afterburner mode. When enabled, quality is increased at
    the expense of additional computational workload.

    :   Off

    :   On (default)

-L, --lowdelay-sbr \<n\>
:   Configure SBR activity on AAC ELD.

    :   Use ELD SBR auto configuration

    :   Disable SBR on ELD (default)

    :   Enable SBR on ELD

-s, --sbr-ratio \<n\>
:   Controls activation of downsampled SBR.

    :   Use lib default (default)

    :   Use downsampled SBR (default for ELD+SBR)

    :   Use dual-rate SBR (default for HE-AAC)

    Dual-rate SBR is what is normally used for HE-AAC, where AAC is
    encoded at half the sample rate of SBR, hence "dual rate". On the
    other hand, downsampled SBR uses same sample rate for both of AAC
    and SBR (single rate), therefore downsampled SBR typically consumes
    more bitrate.

    Downsampled SBR is newly introduced feature in FDK encoder library
    version 3.4.12. When libfdk-aac in the system doesn't support this,
    dual-rate SBR will be used. When available, dual-rate SBR is the
    default for HE-AAC and downsampled SBR is the default for ELD+SBR.

    Note that downsampled HE-AAC is not so common as dual-rate one. When
    downsampled HE-AAC is selected, **fdkaac** is forced to choose
    explicit hierarchical SBR signaling, which (at least) iTunes doesn't

-f, --transport-format \<n\>
:   Transport format. Tagging and gapless playback is only available on
    M4A. Streaming to stdout is only available on others.

    :   M4A (default)

    :   ADIF

    :   ADTS

    :   LATM MCP=1

    :   LATM MCP=0

    :   LOAS/LATM (LATM within LOAS)

-C, --adts-crc-check
:   Add CRC protection on ADTS header.

-h, --header-period \<n\>
:   StreamMuxConfig/PCE repetition period in the transport layer.

-G, --gapless-mode \<n\>
:   Method to declare amount of encoder delay (and padding) in M4A
    container. These values are mandatory for proper gapless playback on
    player side.

    :   iTunSMPB (default)

    :   ISO standard (edts and sgpd)

    :   Both

:   When specified, count SBR decoder delay in encoder delay.

    This is not iTunes compatible and will lead to gapless playback
    issue on LC only decoder, but this is the default behavior of FDK

    Whether counting SBR decoder delay in encoder delay or not result in
    incompatibility in gapless playback. You should pick which one will
    work for your favorite player.

    However, it's better not to choose SBR at all if you want gapless
    playback. LC doesn't have such issues.

-I, --ignorelength
:   Ignore length field of data chunk in input WAV file.

-S, --silent
:   Don't print progress messages.

:   Place moov box before mdat box in M4A container. This option might
    be important for some hardware players, that are known to refuse
    moov box placed after mdat box.

-R, --raw
:   Regard input as raw PCM.

--raw-channels \<n\>
:   Specify number of channels of raw input (default: 2)

--raw-rate \<n\>
:   Specify sample rate of raw input (default: 44100)

--raw-format \<spec\>
:   Specify sample format of raw input (default: "S16L"). **Spec** is as
    the following (case insensitive):

    1st char -- type of sample
    :   **S** (igned) | **U** (nsigned) | **F** (loat)

    2nd part (in digits)
    :   bits per channel

    Last char -- endianness (can be omitted)
    :   **L** (ittle, default) | **B** (ig)

--title \<string\>
:   Set title tag.

--artist \<string\>
:   Set artist tag.

--album \<string\>
:   Set album tag.

--genre \<string\>
:   Set genre tag.

--date \<string\>
:   Set date tag.

--composer \<string\>
:   Set composer tag.

--grouping \<string\>
:   Set grouping tag.

--comment \<string\>
:   Set comment tag.

--album-artist \<string\>
:   Set album artist tag.

--track \<number[/total]\>
:   Set track tag, with or without number of total tracks.

--disk \<number[/total]\>
:   Set disk tag, with or without number of total discs.

--tempo \<n\>
:   Set tempo (BPM) tag.

--tag \<fcc\>:\<value\>
:   Set iTunes predefined tag with explicit fourcc key and value. See
    for known predefined keys. You can omit first char of **fcc** when
    it is the copyright sign.

--tag-from-file \<fcc\>:\<filename\>
:   Same as --tag, but set content of file as tag value.

--long-tag \<name\>:\<value\>
:   Set arbitrary tag as iTunes custom metadata. Stored in field.

--tag-from-json \<filename[?dot\_notation]\>
:   Read tags from JSON. By default, tags are assumed to be direct
    children of the root object in JSON. Optionally you can specify
    arbitrary dot notation to locate the object containing tags.


Encode WAV file into a M4A file. MPEG4 AAC LC, VBR quality 3:

    fdkaac -m3 foo.wav

Encode WAV file into a M4A file. MPEG4 HE-AAC, bitrate 64kbps:

    fdkaac -p5 -b64 foo.wav

Piping from **ffmpeg** (you need version supporting CAF output):

    ffmpeg -i foo.flac -f caf - | fdkaac -b128 - -o foo.m4a

Import tags via json:

    ffprobe -v 0 -of json -show_format foo.flac >foo.json

    flac -dc foo.flac | \
    fdkaac - -ox.m4a -m2 --import-tag-from-json=foo.json?format.tags


Upto 32bit integer or 64bit floating point format is supported as input.
However, FDK library is implemented based on fixed point math and only
supports 16bit integer PCM. Therefore, be wary of clipping. You might
want to dither/noise shape beforehand when your input has higher

Following channel layouts are supported by the encoder.

:   C

:   L R

:   C L R

:   C L R Cs

:   C L R Ls Rs

:   C L R Ls Rs LFE

7.1ch (front)
:   C Lc Rc L R Ls Rs LFE

7.1ch (rear)
:   C L R Ls Rs Rls Rrs LFE