Provided by: fdkaac_0.6.2-1_amd64 bug

NAME

       fdkaac - command line frontend for libfdk-aac encoder

SYNOPSIS

       fdkaac [OPTIONS] [FILE]

DESCRIPTION

       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.

OPTIONS

       -h, --help
              Show command help

       -o <FILE>
              Output filename.

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

              2      MPEG-4 AAC LC (default)

              5      MPEG-4 HE-AAC (SBR)

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

              23     MPEG-4 AAC LD

              39     MPEG-4 AAC ELD

              129    MPEG-2 AAC LC

              132    MPEG-2 HE-AAC (SBR)

              156    MPEG-2 HE-AAC v2 (SBR+PS)

       -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 channels.

              0      CBR (default)

              1-5    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.

              0      Off

              1      On (default)

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

              -1     Use ELD SBR auto configuration

              0      Disable SBR on ELD (default)

              1      Enable SBR on ELD

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

              0      Use lib default (default)

              1      Use downsampled SBR (default for ELD+SBR)

              2      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 accept.

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

              0      M4A (default)

              1      ADIF

              2      ADTS

              6      LATM MCP=1

              7      LATM MCP=0

              10     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.

              0      iTunSMPB (default)

              1      ISO standard (edts and sgpd)

              2      Both

       --include-sbr-delay
              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 library.

              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.

       --moov-before-mdat
              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
              <https://code.google.com/p/mp4v2/wiki/iTunesMetadata> 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 com.apple.iTunes 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.

EXAMPLES

       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

NOTES

       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 resolution.

       Following channel layouts are supported by the encoder.

       1ch    C

       2ch    L R

       3ch    C L R

       4ch    C L R Cs

       5ch    C L R Ls Rs

       5.1ch  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

AUTHORS

       nu774 <honeycomb77@gmail.com>.

                                          November, 2013                                FDKAAC(1)