qaac 2.70 Usage: qaac [options] infiles.... "-" as infile means stdin. On ADTS/WAV output mode, "-" as outfile means stdout. Main options: --formats Show available AAC formats and exit -a, --abr AAC ABR mode / bitrate -V, --tvbr AAC True VBR mode / quality [0-127] -v, --cvbr AAC Constrained VBR mode / bitrate -c, --cbr AAC CBR mode / bitrate For -a, -v, -c, "0" as bitrate means "highest". Highest bitrate available is automatically chosen. For LC, default is -V90 For HE, default is -v0 --he HE AAC mode (TVBR is not available) -q, --quality AAC encoding Quality [0-2] --adts ADTS output (AAC only) --no-smart-padding Don't apply smart padding for gapless playback. By default, beginning and ending of input is extrapolated to achieve smooth transition between songs. This option also works as a workaround for bug of CoreAudio HE-AAC encoder that stops encoding 1 frame too early. Setting this option can lead to gapless playback issue especially on HE-AAC. However, resulting bitstream will be identical with iTunes only when this option is set. -d Output directory. Default is current working dir. --check Show library versions and exit. -A, --alac ALAC encoding mode -D, --decode Decode to a WAV file. --caf Output to CAF file instead of M4A/WAV/AAC. --play Decode to a WaveOut device (playback). -r, --rate keep: output sampling rate will be same as input if possible. auto: output sampling rate will be automatically chosen by encoder. n: desired output sampling rate in Hz. --lowpass Specify lowpass filter cut-off frequency in Hz. Use this when you want lower cut-off than Apple default. -b, --bits-per-sample Bits per sample of output (for WAV/ALAC only) --no-dither Turn off dither when quantizing to lower bit depth. --peak Scan + print peak (don't generate output file). Cannot be used with encoding mode or -D. When DSP options are set, peak is computed after all DSP filters have been applied. --gain Adjust gain by f dB. Use negative value to decrese gain, when you want to avoid clipping introduced by DSP. -N, --normalize Normalize (works in two pass. can generate HUGE tempfile for large piped input) --drc Dynamic range compression. Loud parts over threshold are attenuated by ratio. thresh: threshold (in dBFS, < 0.0) ratio: compression ratio (> 1.0) knee: knee width (in dB, >= 0.0) attack: attack time (in millis, >= 0.0) release: release time (in millis, >= 0.0) --limiter Apply smart limiter that softly clips portions where peak exceeds (near) 0dBFS --start <[[hh:]mm:]ss[.ss..]|s|f> Specify start point of the input. You specify either in seconds(hh:mm:ss.sss..form) or number of samples followed by 's' or cuesheet frames(mm:ss:ff form) followed by 'f'. When negative value is given, instead of trimming, specified amount of silence is prepended. Example: --start 4010160s : start at 4010160 samples --start 1:30:70f : same as above, in cuepoint --start 1:30.93333 : same as above --end <[[hh:]mm:]ss[.ss..]|s|f> Specify end point of the input (exclusive). --delay <[[hh:]mm:]ss[.ss..]|s|f> Same as --start, with the sign reversed. Positive value will prepend silence. (This option exists due to historical reason) --no-delay Compensate encoder delay by prepending 960 samples of scilence, then trimming 3 AAC frames from the beginning (and also tweak iTunSMPB). This option is mainly intended for resolving A/V sync issue of video. --num-priming (Experimental). Set arbitrary number of priming samples in range from 0 to 2112 (default 2112). Applicable only for AAC LC. --num-priming=0 is the same as --no-delay. Doesn't work with --no-smart-padding. --gapless-mode Encoder delay signaling for gapless playback. 0: iTunSMPB (default) 1: ISO standard (elst + sbgp + sgpd) 2: Both --matrix-preset Specify user defined preset for matrix mixer. --matrix-file Matrix file for remix. --no-matrix-normalize Don't automatically normalize(scale) matrix coefficients for the matrix mixer. --chanmap Rearrange input channels to the specified order. Example: --chanmap 2,1 -> swap L and R. --chanmap 2,3,1 -> C+L+R -> L+R+C. --chanmask Force input channel mask(bitmap). Either decimal or hex number with 0x prefix can be used. When 0 is given, qaac works as if no channel mask is present in the source and picks default layout. --no-optimize Don't optimize MP4 container after encoding. --tmpdir Specify temporary directory. Default is %TMP% -s, --silent Suppress console messages. --verbose More verbose console messages. -i, --ignorelength Assume WAV input and ignore the data chunk length. --threading Enable multi-threading. -n, --nice Give lower process priority. --sort-args Sort filenames given by command line arguments. --text-codepage Specify text code page of cuesheet/chapter/lyrics. Example: 1252 for Latin-1, 65001 for UTF-8. Use this when bogus values are written into tags due to automatic encoding detection failure. -S, --stat Save bitrate statistics into file. --log Output message to file. Option for output filename generation: --fname-from-tag Generate filename based on metadata of input. By default, output filename will be the same as input (only different by the file extension). Name generation can be tweaked by --fname-format. --fname-format Format string for output filename. Option for single output: -o Specify output filename --concat Encodes whole inputs into a single file. Requires output filename (with -o) Option for cuesheet input only: --cue-tracks Limit extraction to specified tracks. Tracks can be specified with comma separated numbers. Hyphen can be used to denote range of numbers. Tracks non-existent in the cue are just ignored. Numbers must be in the range 0-99. Example: --cue-tracks 1-3,6-9,11 -> equivalent to --cue-tracks 1,2,3,6,7,8,9,11 --cue-tracks 2-99 -> can be used to skip first track (and HTOA) Options for Raw PCM input only: -R, --raw Raw PCM input. --raw-channels Number of channels, default 2. --raw-rate Sample rate, default 44100. --raw-format Sample format, default S16L. Sample format spec: 1st char: S(igned) | U(nsigned) | F(loat) 2nd part: Bitwidth Last part: L(ittle Endian) | B(ig Endian) Last part can be omitted, L is assumed by default. Cases are ignored. u16b is OK. Options for CoreAudio sample rate converter: --native-resampler[=line|norm|bats,n] Arguments followed by '=' are optional. First argument before comma is complexity from one of the following: line: linear (worst, don't use this) norm: normal bats: mastering (best, default) Second argument after comma is integer quality between 0-127 (default 0). Example: --native-resampler --native-resampler=norm,96 Tagging options: (same value is set to all files, so use with care for multiple files) --title --artist --band This means "Album Artist". --album --grouping --composer --comment --genre --date --track --disk --compilation[=0|1] By default, iTunes compilation flag is not set. --compilation or --compilation=1 sets flag on. --compilation=0 is same as default. --lyrics --artwork --artwork-size Specify maximum width or height of artwork in pixels. If specified artwork (with --artwork) is larger than this, artwork is automatically resized. --copy-artwork Copy front cover art(APIC:type 3) from the source. When --artwork is also given, this option is ignored. --chapter Set chapter from file. --tag : Set iTunes pre-defined tag with fourcc key and value. 1) When key starts with U+00A9 (copyright sign), you can use 3 chars starting from the second char instead. 2) Some known tags having type other than UTF-8 string are taken care of. Others are just stored as UTF-8 string. --tag-from-file : Same as above, but value is read from file. --long-tag : Set long tag (iTunes custom metadata) with arbitrary key/value pair. Value is always stored as UTF8 string.

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