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.TH mpg123 1 "31 Jan 2008"
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.SH NAME
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mpg123 \- play audio MPEG 1.0/2.0/2.5 stream (layers 1, 2 and 3)
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.SH SYNOPSIS
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.B mpg123
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[
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.B options
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]
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.IR file " ... | " URL " ... | "
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.B \-
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.SH DESCRIPTION
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.B mpg123
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reads one or more
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.IR file\^ s
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(or standard input if ``\-'' is specified) or
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.IR URL\^ s
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and plays them on the audio device (default) or
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outputs them to stdout.
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.IR file\^ / URL
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is assumed to be an MPEG audio bit stream.
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.SH OPERANDS
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The following operands are supported:
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.TP 8
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.IR file (s)
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The path name(s) of one or more input files. They must be
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valid MPEG-1.0/2.0/2.5 audio layer 1, 2 or 3 bit streams.
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If a dash ``\-'' is specified, MPEG data will
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be read from the standard input. Furthermore, any name
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starting with ``http://'' is recognized as
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.I URL
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(see next section).
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.SH OPTIONS
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.B mpg123
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options may be either the traditional POSIX one letter options,
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or the GNU style long options. POSIX style options start with a
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single ``\-'', while GNU long options start with ``\-\^\-''.
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Option arguments (if needed) follow separated by whitespace (not ``='').
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Note that some options can be absent from your installation when disabled in the build process.
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.SH INPUT OPTIONS
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.TP
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\fB\-k \fInum\fR, \fB\-\^\-skip \fInum
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Skip first
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.I num
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frames. By default the decoding starts at the first frame.
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.TP
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\fB\-n \fInum\fR, \fB\-\^\-frames \fInum
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Decode only
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.I num
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frames. By default the complete stream is decoded.
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.TP
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.BR \-\-fuzzy
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Enable fuzzy seeks (guessing byte offsets or using approximate seek points from Xing TOC).
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Without that, seeks need a first scan through the file before they can jump at positions.
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You can decide here: sample-accurate operation with gapless features or faster (fuzzy) seeking.
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.TP
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.BR \-y ", " \-\^\-no\-resync
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Do NOT try to resync and continue decoding if an error occurs in
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the input file. Normally,
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.B mpg123
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tries to keep the playback alive at all costs, including skipping invalid material and searching new header when something goes wrong.
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With this switch you can make it bail out on data errors
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(and perhaps spare your ears a bad time). Note that this switch has been renamed from \-\-resync.
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The old name still works, but is not advertised or recommened to use (subject to removal in future).
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.TP
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\fB\-\^-resync\-limit \fIbytes\fR
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Set number of bytes to search for valid MPEG data; <0 means search whole stream.
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If you know there are huge chunks of invalid data in your files... here is your hammer.
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.TP
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\fB\-p \fIURL \fR| \fBnone\fR, \fB\-\^\-proxy \fIURL \fR| \fBnone
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The specified
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.I proxy
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will be used for HTTP requests. It
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should be specified as full URL (``http://host.domain:port/''),
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but the ``http://'' prefix, the port number and the trailing
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slash are optional (the default port is 80). Specifying
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.B none
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means not to use any proxy, and to retrieve files directly
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from the respective servers. See also the
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``HTTP SUPPORT'' section.
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.TP
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\fB\-u \fIauth\fR, \fB\-\^\-auth \fIauth
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HTTP authentication to use when recieving files via HTTP.
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The format used is user:password.
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.TP
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\fB\-@ \fIfile\fR, \fB\-\^\-list \fIfile
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Read filenames and/or URLs of MPEG audio streams from the specified
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.I file
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in addition to the ones specified on the command line (if any).
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Note that
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.I file
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can be either an ordinary file, a dash ``\-'' to indicate that
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a list of filenames/URLs is to be read from the standard input,
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or an URL pointing to a an appropriate list file. Note: only
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one
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.B \-@
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option can be used (if more than one is specified, only the
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last one will be recognized).
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.TP
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\fB\-l \fIn\fR, \fB\-\^\-listentry \fIn
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Of the playlist, play specified entry only.
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.I n
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is the number of entry starting at 1. A value of 0 is the default and means playling the whole list, a negative value means showing of the list of titles with their numbers...
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.TP
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\fB\-\-loop \fItimes\fR
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for looping track(s) a certain number of times, < 0 means infinite loop (not with --random!).
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.TP
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.BR \-\-keep\-open
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For remote control mode: Keep loaded file open after reaching end.
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.TP
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\fB\-\-timeout \fIseconds\fR
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Timeout in (integer) seconds before declaring a stream dead (if <= 0, wait forever).
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.TP
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.BR \-z ", " \-\^\-shuffle
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Shuffle play. Randomly shuffles the order of files specified on the command
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line, or in the list file.
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.TP
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.BR \-Z ", " \-\-random
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Continuous random play. Keeps picking a random file from the command line
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or the play list. Unlike shuffle play above, random play never ends, and
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plays individual songs more than once.
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.TP
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\fB\-\^\-no\-icy\-meta
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Do not accept ICY meta data.
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.TP
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\fB\-i, \-\^-\index
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Index / scan through the track before playback.
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This fills the index table for seeking (if enabled in libmpg123) and may make the operating system cache the file contents for smoother operating on playback.
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.TP
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\fB\-\-index\-size \fIsize\fR
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Set the number of entries in the seek frame index table.
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.TP
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\fB\-\-preframes \fInum\fR
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Set the number of frames to be read as lead-in before a seeked-to position.
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This serves to fill the layer 3 bit reservoir, which is needed to faithfully reproduce a certain sample at a certain position.
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Note that for layer 3, a minimum of 1 is enforced (because of frame overlap), and for layer 1 and 2, this is limited to 2 (no bit reservoir in that case, but engine spin-up anyway).
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.SH OUTPUT and PROCESSING OPTIONS
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.TP
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\fB\-o \fImodule\fR, \-\^\-output \fImodule\fR
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Select audio output module. You can provide a comma-separated list to use the first one that works.
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.TP
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\fB\-\^\-list\-modules
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List the available modules.
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.TP
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\fB\-a \fIdev\fR, \fB\-\^\-audiodevice \fIdev
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Specify the audio device to use. The default is
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system-dependent (usually /dev/audio or /dev/dsp).
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Use this option if you have multiple audio devices and
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the default is not what you want.
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.TP
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.BR \-s ", " \-\^\-stdout
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The decoded audio samples are written to standard output,
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instead of playing them through the audio device. This
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option must be used if your audio hardware is not supported
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by
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.BR mpg123 .
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The output format per default is raw (headerless) linear PCM audio data,
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16 bit, stereo, host byte order (you can force mono or 8bit).
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.TP
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\fB\-O \fIfile\fR, \fB\-\^\-outfile
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Write raw output into a file (instead of simply redirecting standard output to a file with the shell).
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.TP
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\fB\-w \fIfile\fR, \fB\-\^\-wav
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Write output as WAV file. This will cause the MPEG stream to be decoded
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and saved as file
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.I file
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, or standard output if
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.I -
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is used as file name. You can also use
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.I --au
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and
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.I --cdr
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for AU and CDR format, respectively.
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.TP
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\fB\-\^\-au \fIfile
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Does not play the MPEG file but writes it to
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.I file
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in SUN audio format. If \- is used as the filename, the AU file is
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written to stdout.
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.TP
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\fB\-\^\-cdr \fIfile
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Does not play the MPEG file but writes it to
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.I file
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as a CDR file. If \- is used as the filename, the CDR file is written
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to stdout.
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.TP
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.BR \-\-reopen
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Forces reopen of the audiodevice after ever song
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.TP
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.BR \-\-cpu\ \fIdecoder\-type
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Selects a certain decoder (optimized for specific CPU), for example i586 or MMX.
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The list of available decoders can vary; depending on the build and what your CPU supports.
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This options is only availabe when the build actually includes several optimized decoders.
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.TP
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.BR \-\-test\-cpu
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Tests your CPU and prints a list of possible choices for \-\-cpu.
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.TP
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.BR \-\-list\-cpu
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Lists all available decoder choices, regardless of support by your CPU.
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.TP
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\fB\-g \fIgain\fR, \fB\-\^\-gain \fIgain
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[DEPRECATED] Set audio hardware output gain (default: don't change). The unit of the gain value is hardware and output module dependent.
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(This parameter is only provided for backwards compatibility and may be removed in the future without prior notice. Use the audio player for playing and a mixer app for mixing, UNIX style!)
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.TP
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\fB\-f \fIfactor\fR, \fB\-\^\-scale \fIfactor
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Change scale factor (default: 32768).
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.TP
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.BR \-\-rva-mix,\ \-\-rva-radio
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Enable RVA (relative volume adjustment) using the values stored for ReplayGain radio mode / mix mode with all tracks roughly equal loudness.
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The first valid information found in ID3V2 Tags (Comment named RVA or the RVA2 frame) or ReplayGain header in Lame/Info Tag is used.
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.TP
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.BR \-\-rva-album,\ \-\-rva-audiophile
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Enable RVA (relative volume adjustment) using the values stored for ReplayGain audiophile mode / album mode with usually the effect of adjusting album loudness but keeping relative loudness inside album.
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The first valid information found in ID3V2 Tags (Comment named RVA_ALBUM or the RVA2 frame) or ReplayGain header in Lame/Info Tag is used.
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.TP
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.BR \-0 ", " \-\^\-single0 "; " \-1 ", " \-\^\-single1
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Decode only channel 0 (left) or channel 1 (right),
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respectively. These options are available for
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stereo MPEG streams only.
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.TP
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.BR \-m ", " \-\^\-mono ", " \-\^\-mix ", " \-\^\-singlemix
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Mix both channels / decode mono. It takes less
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CPU time than full stereo decoding.
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.TP
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.BR \-\-stereo
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Force stereo output
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.TP
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\fB\-r \fIrate\fR, \fB\-\^\-rate \fIrate
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Set sample rate (default: automatic). You may want to
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change this if you need a constant bitrate independed of
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the mpeg stream rate. mpg123 automagically converts the
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rate. You should then combine this with \-\-stereo or \-\-mono.
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.TP
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.BR \-2 ", " \-\^\-2to1 "; " \-4 ", " \-\^\-4to1
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Performs a downsampling of ratio 2:1 (22 kHz) or 4:1 (11 kHz)
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on the output stream, respectively. Saves some CPU cycles, but
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at least the 4:1 ratio sounds ugly.
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.TP
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.BR \-\-pitch\ \fIvalue
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Set hardware pitch (speedup/down, 0 is neutral; 0.05 is 5%). This changes the output sampling rate, so it only works in the range your audio system/hardware supports.
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.TP
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.BR \-\-8bit
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Forces 8bit output
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.TP
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\fB\-d \fIn\fR, \fB\-\^\-doublespeed \fIn
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Only play every
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.IR n 'th
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frame. This will cause the MPEG stream
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to be played
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.I n
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times faster, which can be used for special
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effects. Can also be combined with the
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.B \-\^\-halfspeed
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option to play 3 out of 4 frames etc. Don't expect great
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sound quality when using this option.
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.TP
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\fB\-h \fIn\fR, \fB\-\^\-halfspeed \fIn
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Play each frame
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.I n
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times. This will cause the MPEG stream
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to be played at
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.IR 1 / n 'th
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speed (n times slower), which can be
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used for special effects. Can also be combined with the
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.B \-\^\-doublespeed
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option to double every third frame or things like that.
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Don't expect great sound quality when using this option.
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.TP
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\fB\-E \fIfile\fR, \fB\-\^\-equalizer
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Enables equalization, taken from
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.IR file .
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The file needs to contain 32 lines of data, additional comment lines may
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be prefixed with
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.IR # .
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Each data line consists of two floating-point entries, separated by
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whitespace. They specify the multipliers for left and right channel of
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a certain frequency band, respectively. The first line corresponds to the
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lowest, the 32nd to the highest frequency band.
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Note that you can control the equalizer interactively with the generic control interface.
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.TP
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\fB\-\^\-gapless
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Enable code that cuts (junk) samples at beginning and end of tracks, enabling gapless transitions between MPEG files when encoder padding and codec delays would prevent it.
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This is enabled per default beginning with mpg123 version 1.0.0 .
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.TP
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\fB\-\^\-no\-gapless
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Disable the gapless code. That gives you MP3 decodings that include encoder delay and padding plus mpg123's decoder delay.
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.TP
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\fB\-D \fIn\fR, \fB\-\-delay \fIn
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Insert a delay of \fIn\fR seconds before each track.
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.TP
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.BR "\-o h" ", " \-\^\-headphones
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Direct audio output to the headphone connector (some hardware only; AIX, HP, SUN).
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.TP
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.BR "\-o s" ", " \-\^\-speaker
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Direct audio output to the speaker (some hardware only; AIX, HP, SUN).
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.TP
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.BR "\-o l" ", " \-\^\-lineout
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Direct audio output to the line-out connector (some hardware only; AIX, HP, SUN).
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.TP
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\fB\-b \fIsize\fR, \fB\-\^\-buffer \fIsize
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Use an audio output buffer of
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.I size
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Kbytes. This is useful to bypass short periods of heavy
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system activity, which would normally cause the audio output
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to be interrupted.
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You should specify a buffer size of at least 1024
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(i.e. 1 Mb, which equals about 6 seconds of audio data) or more;
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less than about 300 does not make much sense. The default is 0,
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which turns buffering off.
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.TP
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\fB\-\^\-preload \fIfraction
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Wait for the buffer to be filled to
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.I fraction
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before starting playback (fraction between 0 and 1). You can tune this prebuffering to either get faster sound to your ears or safer uninterrupted web radio.
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Default is 1 (wait for full buffer before playback).
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.TP
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\fB\-\^\-smooth
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Keep buffer over track boundaries -- meaning, do not empty the buffer between tracks for possibly some added smoothness.
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.SH MISC OPTIONS
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.TP
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.BR \-t ", " \-\^\-test
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Test mode. The audio stream is decoded, but no output occurs.
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.TP
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.BR \-c ", " \-\^\-check
|
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Check for filter range violations (clipping), and report them for each frame
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if any occur.
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.TP
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.BR \-v ", " \-\^\-verbose
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Increase the verbosity level. For example, displays the frame
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numbers during decoding.
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.TP
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.BR \-q ", " \-\^\-quiet
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Quiet. Suppress diagnostic messages.
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.TP
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.BR \-C ", " \-\^\-control
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Enable terminal control keys. By default use 's' or the space bar to stop/restart (pause, unpause) playback, 'f' to jump forward to the next song, 'b' to jump back to the
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beginning of the song, ',' to rewind, '.' to fast forward, and 'q' to quit.
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Type 'h' for a full list of available controls.
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.TP
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\fB\-\^\-title
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In an xterm, or rxvt (compatible, TERM environment variable is examined), change the window's title to the name of song currently
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playing.
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.TP
|
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\fB\-\^\-long\-tag
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Display ID3 tag info always in long format with one line per item (artist, title, ...)
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.TP
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.BR \-\-utf8
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Regardless of environment, print metadata in UTF-8 (otherwise, when not using UTF-8 locale, you'll get ASCII stripdown).
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.TP
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.BR \-R ", " \-\^\-remote
|
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Activate generic control interface.
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.B mpg123
|
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will then read and execute commands from stdin. Basic usage is ``load <filename> '' to play some file and the obvious ``pause'', ``command.
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``jump <frame>'' will jump/seek to a given point (MPEG frame number).
|
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Issue ``help'' to get a full list of commands and syntax.
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.TP
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.BR \-\^\-remote\-err
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Print responses for generic control mode to standard error, not standard out.
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This is automatically triggered when using
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.B -s
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\fN.
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.TP
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\fB\-\-fifo \fIpath
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Create a fifo / named pipe on the given path and use that for reading commands instead of standard input.
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.TP
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\fB\-\^\-aggressive
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Tries to get higher priority
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.TP
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.BR \-T ", " \-\-realtime
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Tries to gain realtime priority. This option usually requires root
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privileges to have any effect.
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.TP
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.BR \-? ", " \-\^\-help
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Shows short usage instructions.
|
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.TP
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.BR \-\^\-longhelp
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Shows long usage instructions.
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.TP
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.BR \-\^\-version
|
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Print the version string.
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.SH HTTP SUPPORT
|
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In addition to reading MPEG audio streams from ordinary
|
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files and from the standard input,
|
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.B mpg123
|
||||
supports retrieval of MPEG audio files or playlists via the HTTP protocol,
|
||||
which is used in the World Wide Web (WWW). Such files are
|
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specified using a so-called URL, which starts with ``http://''. When a file with
|
||||
that prefix is encountered,
|
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.B mpg123
|
||||
attempts to open an HTTP connection to the server in order to
|
||||
retrieve that file to decode and play it.
|
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.P
|
||||
It is often useful to retrieve files through a WWW cache or
|
||||
so-called proxy. To accomplish this,
|
||||
.B mpg123
|
||||
examines the environment for variables named
|
||||
.BR MP3_HTTP_PROXY ", " http_proxy " and " HTTP_PROXY ,
|
||||
in this order. The value of the first one that is set will
|
||||
be used as proxy specification. To override this, you can
|
||||
use the
|
||||
.B \-p
|
||||
command line option (see the ``OPTIONS'' section). Specifying
|
||||
.B "\-p none"
|
||||
will enforce contacting the server directly without using
|
||||
any proxy, even if one of the above environment variables
|
||||
is set.
|
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.P
|
||||
Note that, in order to play MPEG audio files from a WWW
|
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server, it is necessary that the connection to that server
|
||||
is fast enough. For example, a 128 kbit/s MPEG file
|
||||
requires the network connection to be at least 128 kbit/s
|
||||
(16 kbyte/s) plus protocol overhead. If you suffer from
|
||||
short network outages, you should try the
|
||||
.B \-b
|
||||
option (buffer) to bypass such outages. If your network
|
||||
connection is generally not fast enough to retrieve MPEG
|
||||
audio files in realtime, you can first download the files
|
||||
to your local harddisk (e.g. using
|
||||
.BR wget (1))
|
||||
and then play them from there.
|
||||
.P
|
||||
If authentication is needed to access the file it can be
|
||||
specified with the
|
||||
.BR "\-u user:pass".
|
||||
.SH INTERRUPT
|
||||
When in terminal control mode, you can quit via pressing the q key,
|
||||
while any time you can abort
|
||||
.B mpg123
|
||||
by pressing Ctrl-C. If not in terminal control mode, this will
|
||||
skip to the next file (if any). If you want to abort playing immediately
|
||||
in that case, press Ctrl-C twice in short succession (within about one second).
|
||||
.P
|
||||
Note that the result of quitting
|
||||
.B mpg123
|
||||
pressing Ctrl-C might not be audible
|
||||
immediately, due to audio data buffering in the audio device.
|
||||
This delay is system dependent, but it is usually not more
|
||||
than one or two seconds.
|
||||
.SH "SEE ALSO"
|
||||
.BR wget (1),
|
||||
.BR sox (1),
|
||||
.SH NOTES
|
||||
MPEG audio decoding requires a good deal of CPU performance,
|
||||
especially layer-3. To decode it in realtime, you should
|
||||
have at least an i486DX4, Pentium, Alpha, SuperSparc or equivalent
|
||||
processor. You can also use the
|
||||
.B -m
|
||||
option to decode mono only, which reduces the CPU load
|
||||
somewhat for layer-3 streams. See also the
|
||||
.BR \-2 " and " \-4
|
||||
options.
|
||||
.P
|
||||
If everything else fails, use the
|
||||
.B \-s
|
||||
option to decode to standard output, direct it into a file
|
||||
and then use an appropriate utility to play that file.
|
||||
You might have to use a tool such as
|
||||
.BR sox (1)
|
||||
to convert the output to an audio format suitable for
|
||||
your audio player.
|
||||
.P
|
||||
If your system is generally fast enough to decode in
|
||||
realtime, but there are sometimes periods of heavy
|
||||
system load (such as cronjobs, users logging in remotely,
|
||||
starting of ``big'' programs etc.) causing the
|
||||
audio output to be interrupted, then you should use
|
||||
the
|
||||
.B \-b
|
||||
option to use a buffer of reasonable size (at least 1000 Kbytes).
|
||||
.SH BUGS
|
||||
.P
|
||||
Mostly MPEG-1 layer 2 and 3 are tested in real life.
|
||||
Please report any issues and provide test files to help fixing them.
|
||||
.P
|
||||
Free format streams are not supported, but they could be (there is some code).
|
||||
.P
|
||||
No CRC error checking is performed.
|
||||
.P
|
||||
Some platforms lack audio hardware support; you may be able to use the
|
||||
.B -s
|
||||
switch to feed the decoded data to a program that can play it on your audio device.
|
||||
Notably, this includes Tru64 with MME, but you should be able to install and use OSS there (it perhaps will perform better as MME would anyway).
|
||||
.SH AUTHORS
|
||||
.TP
|
||||
Maintainers:
|
||||
.br
|
||||
Thomas Orgis <maintainer@mpg123.org>, <thomas@orgis.org>
|
||||
.br
|
||||
Nicholas J. Humfrey
|
||||
.TP
|
||||
Creator:
|
||||
.br
|
||||
Michael Hipp
|
||||
.TP
|
||||
Uses code or ideas from various people, see the AUTHORS file accompanying the source code.
|
||||
.SH LICENSE
|
||||
.B mpg123
|
||||
is licensed under the GNU Lesser/Library General Public License, LGPL, version 2.1 .
|
||||
.SH WEBSITE
|
||||
http://www.mpg123.org
|
||||
.br
|
||||
http://sourceforge.net/projects/mpg123
|
||||
|
|
@ -1,553 +0,0 @@
|
|||
# OpenAL config file.
|
||||
#
|
||||
# Option blocks may appear multiple times, and duplicated options will take the
|
||||
# last value specified. Environment variables may be specified within option
|
||||
# values, and are automatically substituted when the config file is loaded.
|
||||
# Environment variable names may only contain alpha-numeric characters (a-z,
|
||||
# A-Z, 0-9) and underscores (_), and are prefixed with $. For example,
|
||||
# specifying "$HOME/file.ext" would typically result in something like
|
||||
# "/home/user/file.ext". To specify an actual "$" character, use "$$".
|
||||
#
|
||||
# Device-specific values may be specified by including the device name in the
|
||||
# block name, with "general" replaced by the device name. That is, general
|
||||
# options for the device "Name of Device" would be in the [Name of Device]
|
||||
# block, while ALSA options would be in the [alsa/Name of Device] block.
|
||||
# Options marked as "(global)" are not influenced by the device.
|
||||
#
|
||||
# The system-wide settings can be put in /etc/openal/alsoft.conf and user-
|
||||
# specific override settings in $HOME/.alsoftrc.
|
||||
# For Windows, these settings should go into $AppData\alsoft.ini
|
||||
#
|
||||
# Option and block names are case-senstive. The supplied values are only hints
|
||||
# and may not be honored (though generally it'll try to get as close as
|
||||
# possible). Note: options that are left unset may default to app- or system-
|
||||
# specified values. These are the current available settings:
|
||||
|
||||
##
|
||||
## General stuff
|
||||
##
|
||||
[general]
|
||||
|
||||
## disable-cpu-exts: (global)
|
||||
# Disables use of specialized methods that use specific CPU intrinsics.
|
||||
# Certain methods may utilize CPU extensions for improved performance, and
|
||||
# this option is useful for preventing some or all of those methods from being
|
||||
# used. The available extensions are: sse, sse2, sse3, sse4.1, and neon.
|
||||
# Specifying 'all' disables use of all such specialized methods.
|
||||
#disable-cpu-exts =
|
||||
|
||||
## drivers: (global)
|
||||
# Sets the backend driver list order, comma-seperated. Unknown backends and
|
||||
# duplicated names are ignored. Unlisted backends won't be considered for use
|
||||
# unless the list is ended with a comma (e.g. 'oss,' will try OSS first before
|
||||
# other backends, while 'oss' will try OSS only). Backends prepended with -
|
||||
# won't be considered for use (e.g. '-oss,' will try all available backends
|
||||
# except OSS). An empty list means to try all backends.
|
||||
#drivers =
|
||||
|
||||
## channels:
|
||||
# Sets the output channel configuration. If left unspecified, one will try to
|
||||
# be detected from the system, and defaulting to stereo. The available values
|
||||
# are: mono, stereo, quad, surround51, surround51rear, surround61, surround71,
|
||||
# ambi1, ambi2, ambi3. Note that the ambi* configurations provide ambisonic
|
||||
# channels of the given order (using ACN ordering and SN3D normalization by
|
||||
# default), which need to be decoded to play correctly on speakers.
|
||||
#channels =
|
||||
|
||||
## sample-type:
|
||||
# Sets the output sample type. Currently, all mixing is done with 32-bit float
|
||||
# and converted to the output sample type as needed. Available values are:
|
||||
# int8 - signed 8-bit int
|
||||
# uint8 - unsigned 8-bit int
|
||||
# int16 - signed 16-bit int
|
||||
# uint16 - unsigned 16-bit int
|
||||
# int32 - signed 32-bit int
|
||||
# uint32 - unsigned 32-bit int
|
||||
# float32 - 32-bit float
|
||||
#sample-type = float32
|
||||
|
||||
## frequency:
|
||||
# Sets the output frequency. If left unspecified it will try to detect a
|
||||
# default from the system, otherwise it will default to 44100.
|
||||
#frequency =
|
||||
|
||||
## period_size:
|
||||
# Sets the update period size, in sample frames. This is the number of frames
|
||||
# needed for each mixing update. Acceptable values range between 64 and 8192.
|
||||
# If left unspecified it will default to 1/50th of the frequency (20ms, or 882
|
||||
# for 44100, 960 for 48000, etc).
|
||||
#period_size =
|
||||
|
||||
## periods:
|
||||
# Sets the number of update periods. Higher values create a larger mix ahead,
|
||||
# which helps protect against skips when the CPU is under load, but increases
|
||||
# the delay between a sound getting mixed and being heard. Acceptable values
|
||||
# range between 2 and 16.
|
||||
#periods = 3
|
||||
|
||||
## stereo-mode:
|
||||
# Specifies if stereo output is treated as being headphones or speakers. With
|
||||
# headphones, HRTF or crossfeed filters may be used for better audio quality.
|
||||
# Valid settings are auto, speakers, and headphones.
|
||||
#stereo-mode = auto
|
||||
|
||||
## stereo-encoding:
|
||||
# Specifies the encoding method for non-HRTF stereo output. 'panpot' (default)
|
||||
# uses standard amplitude panning (aka pair-wise, stereo pair, etc) between
|
||||
# -30 and +30 degrees, while 'uhj' creates stereo-compatible two-channel UHJ
|
||||
# output, which encodes some surround sound information into stereo output
|
||||
# that can be decoded with a surround sound receiver. If crossfeed filters are
|
||||
# used, UHJ is disabled.
|
||||
#stereo-encoding = panpot
|
||||
|
||||
## ambi-format:
|
||||
# Specifies the channel order and normalization for the "ambi*" set of channel
|
||||
# configurations. Valid settings are: fuma, ambix (or acn+sn3d), acn+n3d
|
||||
#ambi-format = ambix
|
||||
|
||||
## hrtf:
|
||||
# Controls HRTF processing. These filters provide better spatialization of
|
||||
# sounds while using headphones, but do require a bit more CPU power. While
|
||||
# HRTF is used, the cf_level option is ignored. Setting this to auto (default)
|
||||
# will allow HRTF to be used when headphones are detected or the app requests
|
||||
# it, while setting true or false will forcefully enable or disable HRTF
|
||||
# respectively.
|
||||
#hrtf = auto
|
||||
|
||||
## hrtf-mode:
|
||||
# Specifies the rendering mode for HRTF processing. Setting the mode to full
|
||||
# (default) applies a unique HRIR filter to each source given its relative
|
||||
# location, providing the clearest directional response at the cost of the
|
||||
# highest CPU usage. Setting the mode to ambi1, ambi2, or ambi3 will instead
|
||||
# mix to a first-, second-, or third-order ambisonic buffer respectively, then
|
||||
# decode that buffer with HRTF filters. Ambi1 has the lowest CPU usage,
|
||||
# replacing the per-source HRIR filter for a simple 4-channel panning mix, but
|
||||
# retains full 3D placement at the cost of a more diffuse response. Ambi2 and
|
||||
# ambi3 increasingly improve the directional clarity, at the cost of more CPU
|
||||
# usage (still less than "full", given some number of active sources).
|
||||
#hrtf-mode = full
|
||||
|
||||
## hrtf-size:
|
||||
# Specifies the impulse response size, in samples, for the HRTF filter. Larger
|
||||
# values increase the filter quality, while smaller values reduce processing
|
||||
# cost. A value of 0 (default) uses the full filter size in the dataset, and
|
||||
# the default dataset has a filter size of 32 samples at 44.1khz.
|
||||
#hrtf-size = 0
|
||||
|
||||
## default-hrtf:
|
||||
# Specifies the default HRTF to use. When multiple HRTFs are available, this
|
||||
# determines the preferred one to use if none are specifically requested. Note
|
||||
# that this is the enumerated HRTF name, not necessarily the filename.
|
||||
#default-hrtf =
|
||||
|
||||
## hrtf-paths:
|
||||
# Specifies a comma-separated list of paths containing HRTF data sets. The
|
||||
# format of the files are described in docs/hrtf.txt. The files within the
|
||||
# directories must have the .mhr file extension to be recognized. By default,
|
||||
# OS-dependent data paths will be used. They will also be used if the list
|
||||
# ends with a comma. On Windows this is:
|
||||
# $AppData\openal\hrtf
|
||||
# And on other systems, it's (in order):
|
||||
# $XDG_DATA_HOME/openal/hrtf (defaults to $HOME/.local/share/openal/hrtf)
|
||||
# $XDG_DATA_DIRS/openal/hrtf (defaults to /usr/local/share/openal/hrtf and
|
||||
# /usr/share/openal/hrtf)
|
||||
#hrtf-paths =
|
||||
|
||||
## cf_level:
|
||||
# Sets the crossfeed level for stereo output. Valid values are:
|
||||
# 0 - No crossfeed
|
||||
# 1 - Low crossfeed
|
||||
# 2 - Middle crossfeed
|
||||
# 3 - High crossfeed (virtual speakers are closer to itself)
|
||||
# 4 - Low easy crossfeed
|
||||
# 5 - Middle easy crossfeed
|
||||
# 6 - High easy crossfeed
|
||||
# Users of headphones may want to try various settings. Has no effect on non-
|
||||
# stereo modes.
|
||||
#cf_level = 0
|
||||
|
||||
## resampler: (global)
|
||||
# Selects the default resampler used when mixing sources. Valid values are:
|
||||
# point - nearest sample, no interpolation
|
||||
# linear - extrapolates samples using a linear slope between samples
|
||||
# cubic - extrapolates samples using a Catmull-Rom spline
|
||||
# bsinc12 - extrapolates samples using a band-limited Sinc filter (varying
|
||||
# between 12 and 24 points, with anti-aliasing)
|
||||
# fast_bsinc12 - same as bsinc12, except without interpolation between down-
|
||||
# sampling scales
|
||||
# bsinc24 - extrapolates samples using a band-limited Sinc filter (varying
|
||||
# between 24 and 48 points, with anti-aliasing)
|
||||
# fast_bsinc24 - same as bsinc24, except without interpolation between down-
|
||||
# sampling scales
|
||||
#resampler = linear
|
||||
|
||||
## rt-prio: (global)
|
||||
# Sets real-time priority for the mixing thread. Not all drivers may use this
|
||||
# (eg. PortAudio) as they already control the priority of the mixing thread.
|
||||
# 0 and negative values will disable it. Note that this may constitute a
|
||||
# security risk since a real-time priority thread can indefinitely block
|
||||
# normal-priority threads if it fails to wait. Disable this if it turns out to
|
||||
# be a problem.
|
||||
#rt-prio = 1
|
||||
|
||||
## sources:
|
||||
# Sets the maximum number of allocatable sources. Lower values may help for
|
||||
# systems with apps that try to play more sounds than the CPU can handle.
|
||||
#sources = 256
|
||||
|
||||
## slots:
|
||||
# Sets the maximum number of Auxiliary Effect Slots an app can create. A slot
|
||||
# can use a non-negligible amount of CPU time if an effect is set on it even
|
||||
# if no sources are feeding it, so this may help when apps use more than the
|
||||
# system can handle.
|
||||
#slots = 64
|
||||
|
||||
## sends:
|
||||
# Limits the number of auxiliary sends allowed per source. Setting this higher
|
||||
# than the default has no effect.
|
||||
#sends = 6
|
||||
|
||||
## front-stablizer:
|
||||
# Applies filters to "stablize" front sound imaging. A psychoacoustic method
|
||||
# is used to generate a front-center channel signal from the front-left and
|
||||
# front-right channels, improving the front response by reducing the combing
|
||||
# artifacts and phase errors. Consequently, it will only work with channel
|
||||
# configurations that include front-left, front-right, and front-center.
|
||||
#front-stablizer = false
|
||||
|
||||
## output-limiter:
|
||||
# Applies a gain limiter on the final mixed output. This reduces the volume
|
||||
# when the output samples would otherwise clamp, avoiding excessive clipping
|
||||
# noise.
|
||||
#output-limiter = true
|
||||
|
||||
## dither:
|
||||
# Applies dithering on the final mix, for 8- and 16-bit output by default.
|
||||
# This replaces the distortion created by nearest-value quantization with low-
|
||||
# level whitenoise.
|
||||
#dither = true
|
||||
|
||||
## dither-depth:
|
||||
# Quantization bit-depth for dithered output. A value of 0 (or less) will
|
||||
# match the output sample depth. For int32, uint32, and float32 output, 0 will
|
||||
# disable dithering because they're at or beyond the rendered precision. The
|
||||
# maximum dither depth is 24.
|
||||
#dither-depth = 0
|
||||
|
||||
## volume-adjust:
|
||||
# A global volume adjustment for source output, expressed in decibels. The
|
||||
# value is logarithmic, so +6 will be a scale of (approximately) 2x, +12 will
|
||||
# be a scale of 4x, etc. Similarly, -6 will be x1/2, and -12 is about x1/4. A
|
||||
# value of 0 means no change.
|
||||
#volume-adjust = 0
|
||||
|
||||
## excludefx: (global)
|
||||
# Sets which effects to exclude, preventing apps from using them. This can
|
||||
# help for apps that try to use effects which are too CPU intensive for the
|
||||
# system to handle. Available effects are: eaxreverb,reverb,autowah,chorus,
|
||||
# compressor,distortion,echo,equalizer,flanger,modulator,dedicated,pshifter,
|
||||
# fshifter,vmorpher.
|
||||
#excludefx =
|
||||
|
||||
## default-reverb: (global)
|
||||
# A reverb preset that applies by default to all sources on send 0
|
||||
# (applications that set their own slots on send 0 will override this).
|
||||
# Available presets are: None, Generic, PaddedCell, Room, Bathroom,
|
||||
# Livingroom, Stoneroom, Auditorium, ConcertHall, Cave, Arena, Hangar,
|
||||
# CarpetedHallway, Hallway, StoneCorridor, Alley, Forest, City, Moutains,
|
||||
# Quarry, Plain, ParkingLot, SewerPipe, Underwater, Drugged, Dizzy, Psychotic.
|
||||
#default-reverb =
|
||||
|
||||
## trap-alc-error: (global)
|
||||
# Generates a SIGTRAP signal when an ALC device error is generated, on systems
|
||||
# that support it. This helps when debugging, while trying to find the cause
|
||||
# of a device error. On Windows, a breakpoint exception is generated.
|
||||
#trap-alc-error = false
|
||||
|
||||
## trap-al-error: (global)
|
||||
# Generates a SIGTRAP signal when an AL context error is generated, on systems
|
||||
# that support it. This helps when debugging, while trying to find the cause
|
||||
# of a context error. On Windows, a breakpoint exception is generated.
|
||||
#trap-al-error = false
|
||||
|
||||
##
|
||||
## Ambisonic decoder stuff
|
||||
##
|
||||
[decoder]
|
||||
|
||||
## hq-mode:
|
||||
# Enables a high-quality ambisonic decoder. This mode is capable of frequency-
|
||||
# dependent processing, creating a better reproduction of 3D sound rendering
|
||||
# over surround sound speakers. Enabling this also requires specifying decoder
|
||||
# configuration files for the appropriate speaker configuration you intend to
|
||||
# use (see the quad, surround51, etc options below). Currently, up to third-
|
||||
# order decoding is supported.
|
||||
#hq-mode = true
|
||||
|
||||
## distance-comp:
|
||||
# Enables compensation for the speakers' relative distances to the listener.
|
||||
# This applies the necessary delays and attenuation to make the speakers
|
||||
# behave as though they are all equidistant, which is important for proper
|
||||
# playback of 3D sound rendering. Requires the proper distances to be
|
||||
# specified in the decoder configuration file.
|
||||
#distance-comp = true
|
||||
|
||||
## nfc:
|
||||
# Enables near-field control filters. This simulates and compensates for low-
|
||||
# frequency effects caused by the curvature of nearby sound-waves, which
|
||||
# creates a more realistic perception of sound distance. Note that the effect
|
||||
# may be stronger or weaker than intended if the application doesn't use or
|
||||
# specify an appropriate unit scale, or if incorrect speaker distances are set
|
||||
# in the decoder configuration file.
|
||||
#nfc = false
|
||||
|
||||
## nfc-ref-delay
|
||||
# Specifies the reference delay value for ambisonic output when NFC filters
|
||||
# are enabled. If channels is set to one of the ambi* formats, this option
|
||||
# enables NFC-HOA output with the specified Reference Delay parameter. The
|
||||
# specified value can then be shared with an appropriate NFC-HOA decoder to
|
||||
# reproduce correct near-field effects. Keep in mind that despite being
|
||||
# designed for higher-order ambisonics, this also applies to first-order
|
||||
# output. When left unset, normal output is created with no near-field
|
||||
# simulation. Requires the nfc option to also be enabled.
|
||||
#nfc-ref-delay =
|
||||
|
||||
## quad:
|
||||
# Decoder configuration file for Quadraphonic channel output. See
|
||||
# docs/ambdec.txt for a description of the file format.
|
||||
#quad =
|
||||
|
||||
## surround51:
|
||||
# Decoder configuration file for 5.1 Surround (Side and Rear) channel output.
|
||||
# See docs/ambdec.txt for a description of the file format.
|
||||
#surround51 =
|
||||
|
||||
## surround61:
|
||||
# Decoder configuration file for 6.1 Surround channel output. See
|
||||
# docs/ambdec.txt for a description of the file format.
|
||||
#surround61 =
|
||||
|
||||
## surround71:
|
||||
# Decoder configuration file for 7.1 Surround channel output. See
|
||||
# docs/ambdec.txt for a description of the file format. Note: This can be used
|
||||
# to enable 3D7.1 with the appropriate configuration and speaker placement,
|
||||
# see docs/3D7.1.txt.
|
||||
#surround71 =
|
||||
|
||||
##
|
||||
## Reverb effect stuff (includes EAX reverb)
|
||||
##
|
||||
[reverb]
|
||||
|
||||
## boost: (global)
|
||||
# A global amplification for reverb output, expressed in decibels. The value
|
||||
# is logarithmic, so +6 will be a scale of (approximately) 2x, +12 will be a
|
||||
# scale of 4x, etc. Similarly, -6 will be about half, and -12 about 1/4th. A
|
||||
# value of 0 means no change.
|
||||
#boost = 0
|
||||
|
||||
##
|
||||
## PulseAudio backend stuff
|
||||
##
|
||||
[pulse]
|
||||
|
||||
## spawn-server: (global)
|
||||
# Attempts to autospawn a PulseAudio server whenever needed (initializing the
|
||||
# backend, enumerating devices, etc). Setting autospawn to false in Pulse's
|
||||
# client.conf will still prevent autospawning even if this is set to true.
|
||||
#spawn-server = true
|
||||
|
||||
## allow-moves: (global)
|
||||
# Allows PulseAudio to move active streams to different devices. Note that the
|
||||
# device specifier (seen by applications) will not be updated when this
|
||||
# occurs, and neither will the AL device configuration (sample rate, format,
|
||||
# etc).
|
||||
#allow-moves = true
|
||||
|
||||
## fix-rate:
|
||||
# Specifies whether to match the playback stream's sample rate to the device's
|
||||
# sample rate. Enabling this forces OpenAL Soft to mix sources and effects
|
||||
# directly to the actual output rate, avoiding a second resample pass by the
|
||||
# PulseAudio server.
|
||||
#fix-rate = false
|
||||
|
||||
## adjust-latency:
|
||||
# Attempts to adjust the overall latency of device playback. Note that this
|
||||
# may have adverse effects on the resulting internal buffer sizes and mixing
|
||||
# updates, leading to performance problems and drop-outs. However, if the
|
||||
# PulseAudio server is creating a lot of latency, enabling this may help make
|
||||
# it more manageable.
|
||||
#adjust-latency = false
|
||||
|
||||
##
|
||||
## ALSA backend stuff
|
||||
##
|
||||
[alsa]
|
||||
|
||||
## device: (global)
|
||||
# Sets the device name for the default playback device.
|
||||
#device = default
|
||||
|
||||
## device-prefix: (global)
|
||||
# Sets the prefix used by the discovered (non-default) playback devices. This
|
||||
# will be appended with "CARD=c,DEV=d", where c is the card id and d is the
|
||||
# device index for the requested device name.
|
||||
#device-prefix = plughw:
|
||||
|
||||
## device-prefix-*: (global)
|
||||
# Card- and device-specific prefixes may be used to override the device-prefix
|
||||
# option. The option may specify the card id (eg, device-prefix-NVidia), or
|
||||
# the card id and device index (eg, device-prefix-NVidia-0). The card id is
|
||||
# case-sensitive.
|
||||
#device-prefix- =
|
||||
|
||||
## custom-devices: (global)
|
||||
# Specifies a list of enumerated playback devices and the ALSA devices they
|
||||
# refer to. The list pattern is "Display Name=ALSA device;...". The display
|
||||
# names will be returned for device enumeration, and the ALSA device is the
|
||||
# device name to open for each enumerated device.
|
||||
#custom-devices =
|
||||
|
||||
## capture: (global)
|
||||
# Sets the device name for the default capture device.
|
||||
#capture = default
|
||||
|
||||
## capture-prefix: (global)
|
||||
# Sets the prefix used by the discovered (non-default) capture devices. This
|
||||
# will be appended with "CARD=c,DEV=d", where c is the card id and d is the
|
||||
# device number for the requested device name.
|
||||
#capture-prefix = plughw:
|
||||
|
||||
## capture-prefix-*: (global)
|
||||
# Card- and device-specific prefixes may be used to override the
|
||||
# capture-prefix option. The option may specify the card id (eg,
|
||||
# capture-prefix-NVidia), or the card id and device index (eg,
|
||||
# capture-prefix-NVidia-0). The card id is case-sensitive.
|
||||
#capture-prefix- =
|
||||
|
||||
## custom-captures: (global)
|
||||
# Specifies a list of enumerated capture devices and the ALSA devices they
|
||||
# refer to. The list pattern is "Display Name=ALSA device;...". The display
|
||||
# names will be returned for device enumeration, and the ALSA device is the
|
||||
# device name to open for each enumerated device.
|
||||
#custom-captures =
|
||||
|
||||
## mmap:
|
||||
# Sets whether to try using mmap mode (helps reduce latencies and CPU
|
||||
# consumption). If mmap isn't available, it will automatically fall back to
|
||||
# non-mmap mode. True, yes, on, and non-0 values will attempt to use mmap. 0
|
||||
# and anything else will force mmap off.
|
||||
#mmap = true
|
||||
|
||||
## allow-resampler:
|
||||
# Specifies whether to allow ALSA's built-in resampler. Enabling this will
|
||||
# allow the playback device to be set to a different sample rate than the
|
||||
# actual output, causing ALSA to apply its own resampling pass after OpenAL
|
||||
# Soft resamples and mixes the sources and effects for output.
|
||||
#allow-resampler = false
|
||||
|
||||
##
|
||||
## OSS backend stuff
|
||||
##
|
||||
[oss]
|
||||
|
||||
## device: (global)
|
||||
# Sets the device name for OSS output.
|
||||
#device = /dev/dsp
|
||||
|
||||
## capture: (global)
|
||||
# Sets the device name for OSS capture.
|
||||
#capture = /dev/dsp
|
||||
|
||||
##
|
||||
## Solaris backend stuff
|
||||
##
|
||||
[solaris]
|
||||
|
||||
## device: (global)
|
||||
# Sets the device name for Solaris output.
|
||||
#device = /dev/audio
|
||||
|
||||
##
|
||||
## QSA backend stuff
|
||||
##
|
||||
[qsa]
|
||||
|
||||
##
|
||||
## JACK backend stuff
|
||||
##
|
||||
[jack]
|
||||
|
||||
## spawn-server: (global)
|
||||
# Attempts to autospawn a JACK server whenever needed (initializing the
|
||||
# backend, opening devices, etc).
|
||||
#spawn-server = false
|
||||
|
||||
## custom-devices: (global)
|
||||
# Specifies a list of enumerated devices and the ports they connect to. The
|
||||
# list pattern is "Display Name=ports regex;Display Name=ports regex;...". The
|
||||
# display names will be returned for device enumeration, and the ports regex
|
||||
# is the regular expression to identify the target ports on the server (as
|
||||
# given by the jack_get_ports function) for each enumerated device.
|
||||
#custom-devices =
|
||||
|
||||
## connect-ports:
|
||||
# Attempts to automatically connect the client ports to physical server ports.
|
||||
# Client ports that fail to connect will leave the remaining channels
|
||||
# unconnected and silent (the device format won't change to accommodate).
|
||||
#connect-ports = true
|
||||
|
||||
## buffer-size:
|
||||
# Sets the update buffer size, in samples, that the backend will keep buffered
|
||||
# to handle the server's real-time processing requests. This value must be a
|
||||
# power of 2, or else it will be rounded up to the next power of 2. If it is
|
||||
# less than JACK's buffer update size, it will be clamped. This option may
|
||||
# be useful in case the server's update size is too small and doesn't give the
|
||||
# mixer time to keep enough audio available for the processing requests.
|
||||
#buffer-size = 0
|
||||
|
||||
##
|
||||
## WASAPI backend stuff
|
||||
##
|
||||
[wasapi]
|
||||
|
||||
##
|
||||
## DirectSound backend stuff
|
||||
##
|
||||
[dsound]
|
||||
|
||||
##
|
||||
## Windows Multimedia backend stuff
|
||||
##
|
||||
[winmm]
|
||||
|
||||
##
|
||||
## PortAudio backend stuff
|
||||
##
|
||||
[port]
|
||||
|
||||
## device: (global)
|
||||
# Sets the device index for output. Negative values will use the default as
|
||||
# given by PortAudio itself.
|
||||
#device = -1
|
||||
|
||||
## capture: (global)
|
||||
# Sets the device index for capture. Negative values will use the default as
|
||||
# given by PortAudio itself.
|
||||
#capture = -1
|
||||
|
||||
##
|
||||
## Wave File Writer stuff
|
||||
##
|
||||
[wave]
|
||||
|
||||
## file: (global)
|
||||
# Sets the filename of the wave file to write to. An empty name prevents the
|
||||
# backend from opening, even when explicitly requested.
|
||||
# THIS WILL OVERWRITE EXISTING FILES WITHOUT QUESTION!
|
||||
#file =
|
||||
|
||||
## bformat: (global)
|
||||
# Creates AMB format files using first-order ambisonics instead of a standard
|
||||
# single- or multi-channel .wav file.
|
||||
#bformat = false
|
||||
Binary file not shown.
|
|
@ -1,43 +0,0 @@
|
|||
# AmbDec configuration
|
||||
# Written by Ambisonic Decoder Toolbox, version 8.0
|
||||
|
||||
/description 3D7_2h1v_allrad_5200_rE_max_1_band
|
||||
|
||||
/version 3
|
||||
|
||||
/dec/chan_mask 1bf
|
||||
/dec/freq_bands 1
|
||||
/dec/speakers 7
|
||||
/dec/coeff_scale fuma
|
||||
|
||||
/opt/input_scale fuma
|
||||
/opt/nfeff_comp output
|
||||
/opt/delay_comp on
|
||||
/opt/level_comp on
|
||||
/opt/xover_freq 400.000000
|
||||
/opt/xover_ratio 0.000000
|
||||
|
||||
/speakers/{
|
||||
# id dist azim elev conn
|
||||
#-----------------------------------------------------------------------
|
||||
add_spkr LF 1.500000 51.000000 24.000000
|
||||
add_spkr RF 1.500000 -51.000000 24.000000
|
||||
add_spkr CE 1.500000 0.000000 0.000000
|
||||
add_spkr LB 1.500000 180.000000 55.000000
|
||||
add_spkr RB 1.500000 0.000000 -55.000000
|
||||
add_spkr LS 1.500000 129.000000 -24.000000
|
||||
add_spkr RS 1.500000 -129.000000 -24.000000
|
||||
/}
|
||||
|
||||
/matrix/{
|
||||
order_gain 1.000000 0.774597 0.400000 0.000000
|
||||
add_row 0.325031 0.357638 0.206500 0.234037 0.202440 0.135692 0.116927 -0.098768
|
||||
add_row 0.325036 -0.357619 0.206537 0.234033 -0.202427 -0.135680 0.116934 -0.098768
|
||||
add_row 0.080073 -0.000010 -0.000296 0.155843 -0.000016 -0.000011 -0.000623 0.163306
|
||||
add_row 0.353556 0.000002 0.408453 -0.288377 -0.000004 -0.000003 -0.221039 0.077297
|
||||
add_row 0.325297 0.000008 -0.414018 0.232789 0.000004 0.000003 -0.232940 0.018311
|
||||
add_row 0.353558 0.352704 -0.203542 -0.290124 -0.191868 -0.134582 0.110616 -0.038294
|
||||
add_row 0.353556 -0.352691 -0.203576 -0.290115 0.191871 0.134585 0.110612 -0.038293
|
||||
/}
|
||||
|
||||
/end
|
||||
|
|
@ -1,51 +0,0 @@
|
|||
# AmbDec configuration
|
||||
# Written by Ambisonic Decoder Toolbox, version 8.0
|
||||
|
||||
/description Hexagon_2h0p_pinv_match_rV_max_rE_2_band
|
||||
|
||||
/version 3
|
||||
|
||||
/dec/chan_mask 11b
|
||||
/dec/freq_bands 2
|
||||
/dec/speakers 6
|
||||
/dec/coeff_scale fuma
|
||||
|
||||
/opt/input_scale fuma
|
||||
/opt/nfeff_comp input
|
||||
/opt/delay_comp on
|
||||
/opt/level_comp on
|
||||
/opt/xover_freq 400.000000
|
||||
/opt/xover_ratio 0.000000
|
||||
|
||||
/speakers/{
|
||||
# id dist azim elev conn
|
||||
#-----------------------------------------------------------------------
|
||||
add_spkr LF 1.000000 30.000000 0.000000
|
||||
add_spkr RF 1.000000 -30.000000 0.000000
|
||||
add_spkr RS 1.000000 -90.000000 0.000000
|
||||
add_spkr RB 1.000000 -150.000000 0.000000
|
||||
add_spkr LB 1.000000 150.000000 0.000000
|
||||
add_spkr LS 1.000000 90.000000 0.000000
|
||||
/}
|
||||
|
||||
/lfmatrix/{
|
||||
order_gain 1.000000 1.000000 1.000000 0.000000
|
||||
add_row 0.235702 0.166667 0.288675 0.288675 0.166667
|
||||
add_row 0.235702 -0.166667 0.288675 -0.288675 0.166667
|
||||
add_row 0.235702 -0.333333 0.000000 -0.000000 -0.333333
|
||||
add_row 0.235702 -0.166667 -0.288675 0.288675 0.166667
|
||||
add_row 0.235702 0.166667 -0.288675 -0.288675 0.166667
|
||||
add_row 0.235702 0.333333 0.000000 -0.000000 -0.333333
|
||||
/}
|
||||
|
||||
/hfmatrix/{
|
||||
order_gain 1.414214 1.224745 0.707107 0.000000
|
||||
add_row 0.235702 0.166667 0.288675 0.288675 0.166667
|
||||
add_row 0.235702 -0.166667 0.288675 -0.288675 0.166667
|
||||
add_row 0.235702 -0.333333 0.000000 -0.000000 -0.333333
|
||||
add_row 0.235702 -0.166667 -0.288675 0.288675 0.166667
|
||||
add_row 0.235702 0.166667 -0.288675 -0.288675 0.166667
|
||||
add_row 0.235702 0.333333 0.000000 -0.000000 -0.333333
|
||||
/}
|
||||
|
||||
/end
|
||||
|
|
@ -1,46 +0,0 @@
|
|||
# AmbDec configuration
|
||||
# Written by Ambisonic Decoder Toolbox, version 8.0
|
||||
|
||||
# input channel order: WYXVU
|
||||
|
||||
/description itu50-noCenter_2h0p_allrad_5200_rE_max_1_band
|
||||
|
||||
# Although unused in this configuration, the front-center is declared here so
|
||||
# that an appropriate distance may be set (for proper delaying or attenuating
|
||||
# of dialog and such which feed it directly). It otherwise does not contribute
|
||||
# to positional sound output.
|
||||
|
||||
/version 3
|
||||
|
||||
/dec/chan_mask 11b
|
||||
/dec/freq_bands 1
|
||||
/dec/speakers 5
|
||||
/dec/coeff_scale fuma
|
||||
|
||||
/opt/input_scale fuma
|
||||
/opt/nfeff_comp input
|
||||
/opt/delay_comp on
|
||||
/opt/level_comp on
|
||||
/opt/xover_freq 400.000000
|
||||
/opt/xover_ratio 0.000000
|
||||
|
||||
/speakers/{
|
||||
# id dist azim elev conn
|
||||
#-----------------------------------------------------------------------
|
||||
add_spkr LS 1.000000 110.000000 0.000000 system:playback_3
|
||||
add_spkr LF 1.000000 30.000000 0.000000 system:playback_1
|
||||
add_spkr CE 1.000000 0.000000 0.000000 system:playback_5
|
||||
add_spkr RF 1.000000 -30.000000 0.000000 system:playback_2
|
||||
add_spkr RS 1.000000 -110.000000 0.000000 system:playback_4
|
||||
/}
|
||||
|
||||
/matrix/{
|
||||
order_gain 1.00000000e+00 8.66025404e-01 5.00000000e-01 0.000000
|
||||
add_row 4.70934222e-01 3.78169605e-01 -4.00084750e-01 -8.22264454e-02 -4.43765986e-02
|
||||
add_row 2.66639870e-01 2.55418584e-01 3.32591390e-01 2.82949132e-01 8.16816772e-02
|
||||
add_row 0.00000000e+00 0.00000000e+00 0.00000000e+00 0.00000000e+00 0.00000000e+00
|
||||
add_row 2.66634915e-01 -2.55421639e-01 3.32586482e-01 -2.82947688e-01 8.16782588e-02
|
||||
add_row 4.70935891e-01 -3.78173080e-01 -4.00080588e-01 8.22279700e-02 -4.43716394e-02
|
||||
/}
|
||||
|
||||
/end
|
||||
|
|
@ -1,48 +0,0 @@
|
|||
# AmbDec configuration
|
||||
# Written by Ambisonic Decoder Toolbox, version 8.0
|
||||
|
||||
/description itu50_2h0p_allrad_5200_rE_max_1_band
|
||||
|
||||
/version 3
|
||||
|
||||
/dec/chan_mask 11b
|
||||
/dec/freq_bands 2
|
||||
/dec/speakers 5
|
||||
/dec/coeff_scale fuma
|
||||
|
||||
/opt/input_scale fuma
|
||||
/opt/nfeff_comp output
|
||||
/opt/delay_comp on
|
||||
/opt/level_comp on
|
||||
/opt/xover_freq 400.000000
|
||||
/opt/xover_ratio 3.000000
|
||||
|
||||
/speakers/{
|
||||
# id dist azim elev conn
|
||||
#-----------------------------------------------------------------------
|
||||
add_spkr LS 1.000000 110.000000 0.000000
|
||||
add_spkr LF 1.000000 30.000000 0.000000
|
||||
add_spkr CE 1.000000 0.000000 0.000000
|
||||
add_spkr RF 1.000000 -30.000000 0.000000
|
||||
add_spkr RS 1.000000 -110.000000 0.000000
|
||||
/}
|
||||
|
||||
/lfmatrix/{
|
||||
order_gain 1.000000 1.000000 1.000000 0.000000
|
||||
add_row 0.420330 0.330200 -0.312250 0.019350 -0.027010
|
||||
add_row 0.197700 0.288820 0.287820 0.049110 0.007420
|
||||
add_row 0.058030 0.000000 0.205970 0.000000 0.050790
|
||||
add_row 0.197700 -0.288820 0.287820 -0.049110 0.007420
|
||||
add_row 0.420330 -0.330200 -0.312250 -0.019350 -0.027010
|
||||
/}
|
||||
|
||||
/hfmatrix/{
|
||||
order_gain 1.000000 0.866025 0.500000 0.000000
|
||||
add_row 0.470934 0.378170 -0.400085 -0.082226 -0.044377
|
||||
add_row 0.208954 0.257988 0.230383 0.288520 -0.025085
|
||||
add_row 0.109403 -0.000002 0.194278 -0.000003 0.200863
|
||||
add_row 0.208950 -0.257989 0.230379 -0.288516 -0.025088
|
||||
add_row 0.470936 -0.378173 -0.400081 0.082228 -0.044372
|
||||
/}
|
||||
|
||||
/end
|
||||
|
|
@ -1,42 +0,0 @@
|
|||
Ambisonic decoder configuration presets are provided here for common surround
|
||||
sound speaker layouts. The presets are prepared to work with OpenAL Soft's high
|
||||
quality decoder. By default all of the speaker distances within a preset are
|
||||
set to the same value, which results in no effect from distance compensation.
|
||||
If this doesn't match your physical speaker setup, it may be worth copying the
|
||||
preset and modifying the distance values to match (note that modifying the
|
||||
azimuth and elevation values in the presets will not have any effect; the
|
||||
specified angles do not change the decoder behavior).
|
||||
|
||||
Details of the individual presets are as follows.
|
||||
|
||||
square.ambdec
|
||||
Specifies a basic square speaker setup for Quadraphonic output, with identical
|
||||
width and depth. Front speakers are placed at +45 and -45 degrees, and back
|
||||
speakers are placed at +135 and -135 degrees.
|
||||
|
||||
rectangle.ambdec
|
||||
Specifies a narrower speaker setup for Quadraphonic output, with a little less
|
||||
width but a little more depth over a basic square setup. Front speakers are
|
||||
placed at +30 and -30 degrees, providing a bit more compatibility for existing
|
||||
stereo content, with back speakers at +150 and -150 degrees.
|
||||
|
||||
itu5.1.ambdec
|
||||
Specifies a standard ITU 5.0/5.1 setup for 5.1 Surround output. The front-
|
||||
center speaker is placed directly in front at 0 degrees, with the front-left
|
||||
and front-right at +30 and -30 degrees, and the surround speakers (side or
|
||||
back) at +110 and -110 degrees.
|
||||
|
||||
hexagon.ambdec
|
||||
Specifies a flat-front hexagonal speaker setup for 7.1 Surround output. The
|
||||
front left and right speakers are placed at +30 and -30 degrees, the side
|
||||
speakers are placed at +90 and -90 degrees, and the back speakers are placed at
|
||||
+150 and -150 degrees. Although this is for 7.1 output, no front-center speaker
|
||||
is defined for the decoder, meaning that speaker will be silent for 3D sound
|
||||
(however it may still be used with AL_SOFT_direct_channels or ALC_EXT_DEDICATED
|
||||
output). A "proper" 7.1 decoder may be provided in the future, but due to the
|
||||
nature of the speaker configuration will have trade-offs.
|
||||
|
||||
3D7.1.ambdec
|
||||
Specifies a 3D7.1 speaker setup for 7.1 Surround output. Although it's for 7.1
|
||||
output, the speakers for such a configuration need to be placed in different
|
||||
positions for proper results. Please see docs/3D7.1.txt for more information.
|
||||
|
|
@ -1,45 +0,0 @@
|
|||
# AmbDec configuration
|
||||
# Written by Ambisonic Decoder Toolbox, version 8.0
|
||||
|
||||
/description Rectangle_1h0p_pinv_match_rV_max_rE_2_band
|
||||
|
||||
/version 3
|
||||
|
||||
/dec/chan_mask b
|
||||
/dec/freq_bands 2
|
||||
/dec/speakers 4
|
||||
/dec/coeff_scale fuma
|
||||
|
||||
/opt/input_scale fuma
|
||||
/opt/nfeff_comp input
|
||||
/opt/delay_comp on
|
||||
/opt/level_comp on
|
||||
/opt/xover_freq 400.000000
|
||||
/opt/xover_ratio 0.000000
|
||||
|
||||
/speakers/{
|
||||
# id dist azim elev conn
|
||||
#-----------------------------------------------------------------------
|
||||
add_spkr LF 1.000000 30.000000 0.000000
|
||||
add_spkr RF 1.000000 -30.000000 0.000000
|
||||
add_spkr RB 1.000000 -150.000000 0.000000
|
||||
add_spkr LB 1.000000 150.000000 0.000000
|
||||
/}
|
||||
|
||||
/lfmatrix/{
|
||||
order_gain 1.000000 1.000000 0.000000 0.000000
|
||||
add_row 0.353553 0.500000 0.288675
|
||||
add_row 0.353553 -0.500000 0.288675
|
||||
add_row 0.353553 -0.500000 -0.288675
|
||||
add_row 0.353553 0.500000 -0.288675
|
||||
/}
|
||||
|
||||
/hfmatrix/{
|
||||
order_gain 1.414214 1.000000 0.000000 0.000000
|
||||
add_row 0.353553 0.500000 0.288675
|
||||
add_row 0.353553 -0.500000 0.288675
|
||||
add_row 0.353553 -0.500000 -0.288675
|
||||
add_row 0.353553 0.500000 -0.288675
|
||||
/}
|
||||
|
||||
/end
|
||||
|
|
@ -1,45 +0,0 @@
|
|||
# AmbDec configuration
|
||||
# Written by Ambisonic Decoder Toolbox, version 8.0
|
||||
|
||||
/description Square_1h0p_pinv_match_rV_max_rE_2_band
|
||||
|
||||
/version 3
|
||||
|
||||
/dec/chan_mask b
|
||||
/dec/freq_bands 2
|
||||
/dec/speakers 4
|
||||
/dec/coeff_scale fuma
|
||||
|
||||
/opt/input_scale fuma
|
||||
/opt/nfeff_comp input
|
||||
/opt/delay_comp on
|
||||
/opt/level_comp on
|
||||
/opt/xover_freq 400.000000
|
||||
/opt/xover_ratio 0.000000
|
||||
|
||||
/speakers/{
|
||||
# id dist azim elev conn
|
||||
#-----------------------------------------------------------------------
|
||||
add_spkr LF 1.000000 45.000000 0.000000
|
||||
add_spkr RF 1.000000 -45.000000 0.000000
|
||||
add_spkr RB 1.000000 -135.000000 0.000000
|
||||
add_spkr LB 1.000000 135.000000 0.000000
|
||||
/}
|
||||
|
||||
/lfmatrix/{
|
||||
order_gain 1.000000 1.000000 0.000000 0.000000
|
||||
add_row 0.353553 0.353553 0.353553
|
||||
add_row 0.353553 -0.353553 0.353553
|
||||
add_row 0.353553 -0.353553 -0.353553
|
||||
add_row 0.353553 0.353553 -0.353553
|
||||
/}
|
||||
|
||||
/hfmatrix/{
|
||||
order_gain 1.414214 1.000000 0.000000 0.000000
|
||||
add_row 0.353553 0.353553 0.353553
|
||||
add_row 0.353553 -0.353553 0.353553
|
||||
add_row 0.353553 -0.353553 -0.353553
|
||||
add_row 0.353553 0.353553 -0.353553
|
||||
/}
|
||||
|
||||
/end
|
||||
|
|
@ -1,553 +0,0 @@
|
|||
# OpenAL config file.
|
||||
#
|
||||
# Option blocks may appear multiple times, and duplicated options will take the
|
||||
# last value specified. Environment variables may be specified within option
|
||||
# values, and are automatically substituted when the config file is loaded.
|
||||
# Environment variable names may only contain alpha-numeric characters (a-z,
|
||||
# A-Z, 0-9) and underscores (_), and are prefixed with $. For example,
|
||||
# specifying "$HOME/file.ext" would typically result in something like
|
||||
# "/home/user/file.ext". To specify an actual "$" character, use "$$".
|
||||
#
|
||||
# Device-specific values may be specified by including the device name in the
|
||||
# block name, with "general" replaced by the device name. That is, general
|
||||
# options for the device "Name of Device" would be in the [Name of Device]
|
||||
# block, while ALSA options would be in the [alsa/Name of Device] block.
|
||||
# Options marked as "(global)" are not influenced by the device.
|
||||
#
|
||||
# The system-wide settings can be put in /etc/openal/alsoft.conf and user-
|
||||
# specific override settings in $HOME/.alsoftrc.
|
||||
# For Windows, these settings should go into $AppData\alsoft.ini
|
||||
#
|
||||
# Option and block names are case-senstive. The supplied values are only hints
|
||||
# and may not be honored (though generally it'll try to get as close as
|
||||
# possible). Note: options that are left unset may default to app- or system-
|
||||
# specified values. These are the current available settings:
|
||||
|
||||
##
|
||||
## General stuff
|
||||
##
|
||||
[general]
|
||||
|
||||
## disable-cpu-exts: (global)
|
||||
# Disables use of specialized methods that use specific CPU intrinsics.
|
||||
# Certain methods may utilize CPU extensions for improved performance, and
|
||||
# this option is useful for preventing some or all of those methods from being
|
||||
# used. The available extensions are: sse, sse2, sse3, sse4.1, and neon.
|
||||
# Specifying 'all' disables use of all such specialized methods.
|
||||
#disable-cpu-exts =
|
||||
|
||||
## drivers: (global)
|
||||
# Sets the backend driver list order, comma-seperated. Unknown backends and
|
||||
# duplicated names are ignored. Unlisted backends won't be considered for use
|
||||
# unless the list is ended with a comma (e.g. 'oss,' will try OSS first before
|
||||
# other backends, while 'oss' will try OSS only). Backends prepended with -
|
||||
# won't be considered for use (e.g. '-oss,' will try all available backends
|
||||
# except OSS). An empty list means to try all backends.
|
||||
#drivers =
|
||||
|
||||
## channels:
|
||||
# Sets the output channel configuration. If left unspecified, one will try to
|
||||
# be detected from the system, and defaulting to stereo. The available values
|
||||
# are: mono, stereo, quad, surround51, surround51rear, surround61, surround71,
|
||||
# ambi1, ambi2, ambi3. Note that the ambi* configurations provide ambisonic
|
||||
# channels of the given order (using ACN ordering and SN3D normalization by
|
||||
# default), which need to be decoded to play correctly on speakers.
|
||||
#channels =
|
||||
|
||||
## sample-type:
|
||||
# Sets the output sample type. Currently, all mixing is done with 32-bit float
|
||||
# and converted to the output sample type as needed. Available values are:
|
||||
# int8 - signed 8-bit int
|
||||
# uint8 - unsigned 8-bit int
|
||||
# int16 - signed 16-bit int
|
||||
# uint16 - unsigned 16-bit int
|
||||
# int32 - signed 32-bit int
|
||||
# uint32 - unsigned 32-bit int
|
||||
# float32 - 32-bit float
|
||||
#sample-type = float32
|
||||
|
||||
## frequency:
|
||||
# Sets the output frequency. If left unspecified it will try to detect a
|
||||
# default from the system, otherwise it will default to 44100.
|
||||
#frequency =
|
||||
|
||||
## period_size:
|
||||
# Sets the update period size, in sample frames. This is the number of frames
|
||||
# needed for each mixing update. Acceptable values range between 64 and 8192.
|
||||
# If left unspecified it will default to 1/50th of the frequency (20ms, or 882
|
||||
# for 44100, 960 for 48000, etc).
|
||||
#period_size =
|
||||
|
||||
## periods:
|
||||
# Sets the number of update periods. Higher values create a larger mix ahead,
|
||||
# which helps protect against skips when the CPU is under load, but increases
|
||||
# the delay between a sound getting mixed and being heard. Acceptable values
|
||||
# range between 2 and 16.
|
||||
#periods = 3
|
||||
|
||||
## stereo-mode:
|
||||
# Specifies if stereo output is treated as being headphones or speakers. With
|
||||
# headphones, HRTF or crossfeed filters may be used for better audio quality.
|
||||
# Valid settings are auto, speakers, and headphones.
|
||||
#stereo-mode = auto
|
||||
|
||||
## stereo-encoding:
|
||||
# Specifies the encoding method for non-HRTF stereo output. 'panpot' (default)
|
||||
# uses standard amplitude panning (aka pair-wise, stereo pair, etc) between
|
||||
# -30 and +30 degrees, while 'uhj' creates stereo-compatible two-channel UHJ
|
||||
# output, which encodes some surround sound information into stereo output
|
||||
# that can be decoded with a surround sound receiver. If crossfeed filters are
|
||||
# used, UHJ is disabled.
|
||||
#stereo-encoding = panpot
|
||||
|
||||
## ambi-format:
|
||||
# Specifies the channel order and normalization for the "ambi*" set of channel
|
||||
# configurations. Valid settings are: fuma, ambix (or acn+sn3d), acn+n3d
|
||||
#ambi-format = ambix
|
||||
|
||||
## hrtf:
|
||||
# Controls HRTF processing. These filters provide better spatialization of
|
||||
# sounds while using headphones, but do require a bit more CPU power. While
|
||||
# HRTF is used, the cf_level option is ignored. Setting this to auto (default)
|
||||
# will allow HRTF to be used when headphones are detected or the app requests
|
||||
# it, while setting true or false will forcefully enable or disable HRTF
|
||||
# respectively.
|
||||
#hrtf = auto
|
||||
|
||||
## hrtf-mode:
|
||||
# Specifies the rendering mode for HRTF processing. Setting the mode to full
|
||||
# (default) applies a unique HRIR filter to each source given its relative
|
||||
# location, providing the clearest directional response at the cost of the
|
||||
# highest CPU usage. Setting the mode to ambi1, ambi2, or ambi3 will instead
|
||||
# mix to a first-, second-, or third-order ambisonic buffer respectively, then
|
||||
# decode that buffer with HRTF filters. Ambi1 has the lowest CPU usage,
|
||||
# replacing the per-source HRIR filter for a simple 4-channel panning mix, but
|
||||
# retains full 3D placement at the cost of a more diffuse response. Ambi2 and
|
||||
# ambi3 increasingly improve the directional clarity, at the cost of more CPU
|
||||
# usage (still less than "full", given some number of active sources).
|
||||
#hrtf-mode = full
|
||||
|
||||
## hrtf-size:
|
||||
# Specifies the impulse response size, in samples, for the HRTF filter. Larger
|
||||
# values increase the filter quality, while smaller values reduce processing
|
||||
# cost. A value of 0 (default) uses the full filter size in the dataset, and
|
||||
# the default dataset has a filter size of 32 samples at 44.1khz.
|
||||
#hrtf-size = 0
|
||||
|
||||
## default-hrtf:
|
||||
# Specifies the default HRTF to use. When multiple HRTFs are available, this
|
||||
# determines the preferred one to use if none are specifically requested. Note
|
||||
# that this is the enumerated HRTF name, not necessarily the filename.
|
||||
#default-hrtf =
|
||||
|
||||
## hrtf-paths:
|
||||
# Specifies a comma-separated list of paths containing HRTF data sets. The
|
||||
# format of the files are described in docs/hrtf.txt. The files within the
|
||||
# directories must have the .mhr file extension to be recognized. By default,
|
||||
# OS-dependent data paths will be used. They will also be used if the list
|
||||
# ends with a comma. On Windows this is:
|
||||
# $AppData\openal\hrtf
|
||||
# And on other systems, it's (in order):
|
||||
# $XDG_DATA_HOME/openal/hrtf (defaults to $HOME/.local/share/openal/hrtf)
|
||||
# $XDG_DATA_DIRS/openal/hrtf (defaults to /usr/local/share/openal/hrtf and
|
||||
# /usr/share/openal/hrtf)
|
||||
#hrtf-paths =
|
||||
|
||||
## cf_level:
|
||||
# Sets the crossfeed level for stereo output. Valid values are:
|
||||
# 0 - No crossfeed
|
||||
# 1 - Low crossfeed
|
||||
# 2 - Middle crossfeed
|
||||
# 3 - High crossfeed (virtual speakers are closer to itself)
|
||||
# 4 - Low easy crossfeed
|
||||
# 5 - Middle easy crossfeed
|
||||
# 6 - High easy crossfeed
|
||||
# Users of headphones may want to try various settings. Has no effect on non-
|
||||
# stereo modes.
|
||||
#cf_level = 0
|
||||
|
||||
## resampler: (global)
|
||||
# Selects the default resampler used when mixing sources. Valid values are:
|
||||
# point - nearest sample, no interpolation
|
||||
# linear - extrapolates samples using a linear slope between samples
|
||||
# cubic - extrapolates samples using a Catmull-Rom spline
|
||||
# bsinc12 - extrapolates samples using a band-limited Sinc filter (varying
|
||||
# between 12 and 24 points, with anti-aliasing)
|
||||
# fast_bsinc12 - same as bsinc12, except without interpolation between down-
|
||||
# sampling scales
|
||||
# bsinc24 - extrapolates samples using a band-limited Sinc filter (varying
|
||||
# between 24 and 48 points, with anti-aliasing)
|
||||
# fast_bsinc24 - same as bsinc24, except without interpolation between down-
|
||||
# sampling scales
|
||||
#resampler = linear
|
||||
|
||||
## rt-prio: (global)
|
||||
# Sets real-time priority for the mixing thread. Not all drivers may use this
|
||||
# (eg. PortAudio) as they already control the priority of the mixing thread.
|
||||
# 0 and negative values will disable it. Note that this may constitute a
|
||||
# security risk since a real-time priority thread can indefinitely block
|
||||
# normal-priority threads if it fails to wait. Disable this if it turns out to
|
||||
# be a problem.
|
||||
#rt-prio = 1
|
||||
|
||||
## sources:
|
||||
# Sets the maximum number of allocatable sources. Lower values may help for
|
||||
# systems with apps that try to play more sounds than the CPU can handle.
|
||||
#sources = 256
|
||||
|
||||
## slots:
|
||||
# Sets the maximum number of Auxiliary Effect Slots an app can create. A slot
|
||||
# can use a non-negligible amount of CPU time if an effect is set on it even
|
||||
# if no sources are feeding it, so this may help when apps use more than the
|
||||
# system can handle.
|
||||
#slots = 64
|
||||
|
||||
## sends:
|
||||
# Limits the number of auxiliary sends allowed per source. Setting this higher
|
||||
# than the default has no effect.
|
||||
#sends = 6
|
||||
|
||||
## front-stablizer:
|
||||
# Applies filters to "stablize" front sound imaging. A psychoacoustic method
|
||||
# is used to generate a front-center channel signal from the front-left and
|
||||
# front-right channels, improving the front response by reducing the combing
|
||||
# artifacts and phase errors. Consequently, it will only work with channel
|
||||
# configurations that include front-left, front-right, and front-center.
|
||||
#front-stablizer = false
|
||||
|
||||
## output-limiter:
|
||||
# Applies a gain limiter on the final mixed output. This reduces the volume
|
||||
# when the output samples would otherwise clamp, avoiding excessive clipping
|
||||
# noise.
|
||||
#output-limiter = true
|
||||
|
||||
## dither:
|
||||
# Applies dithering on the final mix, for 8- and 16-bit output by default.
|
||||
# This replaces the distortion created by nearest-value quantization with low-
|
||||
# level whitenoise.
|
||||
#dither = true
|
||||
|
||||
## dither-depth:
|
||||
# Quantization bit-depth for dithered output. A value of 0 (or less) will
|
||||
# match the output sample depth. For int32, uint32, and float32 output, 0 will
|
||||
# disable dithering because they're at or beyond the rendered precision. The
|
||||
# maximum dither depth is 24.
|
||||
#dither-depth = 0
|
||||
|
||||
## volume-adjust:
|
||||
# A global volume adjustment for source output, expressed in decibels. The
|
||||
# value is logarithmic, so +6 will be a scale of (approximately) 2x, +12 will
|
||||
# be a scale of 4x, etc. Similarly, -6 will be x1/2, and -12 is about x1/4. A
|
||||
# value of 0 means no change.
|
||||
#volume-adjust = 0
|
||||
|
||||
## excludefx: (global)
|
||||
# Sets which effects to exclude, preventing apps from using them. This can
|
||||
# help for apps that try to use effects which are too CPU intensive for the
|
||||
# system to handle. Available effects are: eaxreverb,reverb,autowah,chorus,
|
||||
# compressor,distortion,echo,equalizer,flanger,modulator,dedicated,pshifter,
|
||||
# fshifter,vmorpher.
|
||||
#excludefx =
|
||||
|
||||
## default-reverb: (global)
|
||||
# A reverb preset that applies by default to all sources on send 0
|
||||
# (applications that set their own slots on send 0 will override this).
|
||||
# Available presets are: None, Generic, PaddedCell, Room, Bathroom,
|
||||
# Livingroom, Stoneroom, Auditorium, ConcertHall, Cave, Arena, Hangar,
|
||||
# CarpetedHallway, Hallway, StoneCorridor, Alley, Forest, City, Moutains,
|
||||
# Quarry, Plain, ParkingLot, SewerPipe, Underwater, Drugged, Dizzy, Psychotic.
|
||||
#default-reverb =
|
||||
|
||||
## trap-alc-error: (global)
|
||||
# Generates a SIGTRAP signal when an ALC device error is generated, on systems
|
||||
# that support it. This helps when debugging, while trying to find the cause
|
||||
# of a device error. On Windows, a breakpoint exception is generated.
|
||||
#trap-alc-error = false
|
||||
|
||||
## trap-al-error: (global)
|
||||
# Generates a SIGTRAP signal when an AL context error is generated, on systems
|
||||
# that support it. This helps when debugging, while trying to find the cause
|
||||
# of a context error. On Windows, a breakpoint exception is generated.
|
||||
#trap-al-error = false
|
||||
|
||||
##
|
||||
## Ambisonic decoder stuff
|
||||
##
|
||||
[decoder]
|
||||
|
||||
## hq-mode:
|
||||
# Enables a high-quality ambisonic decoder. This mode is capable of frequency-
|
||||
# dependent processing, creating a better reproduction of 3D sound rendering
|
||||
# over surround sound speakers. Enabling this also requires specifying decoder
|
||||
# configuration files for the appropriate speaker configuration you intend to
|
||||
# use (see the quad, surround51, etc options below). Currently, up to third-
|
||||
# order decoding is supported.
|
||||
#hq-mode = true
|
||||
|
||||
## distance-comp:
|
||||
# Enables compensation for the speakers' relative distances to the listener.
|
||||
# This applies the necessary delays and attenuation to make the speakers
|
||||
# behave as though they are all equidistant, which is important for proper
|
||||
# playback of 3D sound rendering. Requires the proper distances to be
|
||||
# specified in the decoder configuration file.
|
||||
#distance-comp = true
|
||||
|
||||
## nfc:
|
||||
# Enables near-field control filters. This simulates and compensates for low-
|
||||
# frequency effects caused by the curvature of nearby sound-waves, which
|
||||
# creates a more realistic perception of sound distance. Note that the effect
|
||||
# may be stronger or weaker than intended if the application doesn't use or
|
||||
# specify an appropriate unit scale, or if incorrect speaker distances are set
|
||||
# in the decoder configuration file.
|
||||
#nfc = false
|
||||
|
||||
## nfc-ref-delay
|
||||
# Specifies the reference delay value for ambisonic output when NFC filters
|
||||
# are enabled. If channels is set to one of the ambi* formats, this option
|
||||
# enables NFC-HOA output with the specified Reference Delay parameter. The
|
||||
# specified value can then be shared with an appropriate NFC-HOA decoder to
|
||||
# reproduce correct near-field effects. Keep in mind that despite being
|
||||
# designed for higher-order ambisonics, this also applies to first-order
|
||||
# output. When left unset, normal output is created with no near-field
|
||||
# simulation. Requires the nfc option to also be enabled.
|
||||
#nfc-ref-delay =
|
||||
|
||||
## quad:
|
||||
# Decoder configuration file for Quadraphonic channel output. See
|
||||
# docs/ambdec.txt for a description of the file format.
|
||||
#quad =
|
||||
|
||||
## surround51:
|
||||
# Decoder configuration file for 5.1 Surround (Side and Rear) channel output.
|
||||
# See docs/ambdec.txt for a description of the file format.
|
||||
#surround51 =
|
||||
|
||||
## surround61:
|
||||
# Decoder configuration file for 6.1 Surround channel output. See
|
||||
# docs/ambdec.txt for a description of the file format.
|
||||
#surround61 =
|
||||
|
||||
## surround71:
|
||||
# Decoder configuration file for 7.1 Surround channel output. See
|
||||
# docs/ambdec.txt for a description of the file format. Note: This can be used
|
||||
# to enable 3D7.1 with the appropriate configuration and speaker placement,
|
||||
# see docs/3D7.1.txt.
|
||||
#surround71 =
|
||||
|
||||
##
|
||||
## Reverb effect stuff (includes EAX reverb)
|
||||
##
|
||||
[reverb]
|
||||
|
||||
## boost: (global)
|
||||
# A global amplification for reverb output, expressed in decibels. The value
|
||||
# is logarithmic, so +6 will be a scale of (approximately) 2x, +12 will be a
|
||||
# scale of 4x, etc. Similarly, -6 will be about half, and -12 about 1/4th. A
|
||||
# value of 0 means no change.
|
||||
#boost = 0
|
||||
|
||||
##
|
||||
## PulseAudio backend stuff
|
||||
##
|
||||
[pulse]
|
||||
|
||||
## spawn-server: (global)
|
||||
# Attempts to autospawn a PulseAudio server whenever needed (initializing the
|
||||
# backend, enumerating devices, etc). Setting autospawn to false in Pulse's
|
||||
# client.conf will still prevent autospawning even if this is set to true.
|
||||
#spawn-server = true
|
||||
|
||||
## allow-moves: (global)
|
||||
# Allows PulseAudio to move active streams to different devices. Note that the
|
||||
# device specifier (seen by applications) will not be updated when this
|
||||
# occurs, and neither will the AL device configuration (sample rate, format,
|
||||
# etc).
|
||||
#allow-moves = true
|
||||
|
||||
## fix-rate:
|
||||
# Specifies whether to match the playback stream's sample rate to the device's
|
||||
# sample rate. Enabling this forces OpenAL Soft to mix sources and effects
|
||||
# directly to the actual output rate, avoiding a second resample pass by the
|
||||
# PulseAudio server.
|
||||
#fix-rate = false
|
||||
|
||||
## adjust-latency:
|
||||
# Attempts to adjust the overall latency of device playback. Note that this
|
||||
# may have adverse effects on the resulting internal buffer sizes and mixing
|
||||
# updates, leading to performance problems and drop-outs. However, if the
|
||||
# PulseAudio server is creating a lot of latency, enabling this may help make
|
||||
# it more manageable.
|
||||
#adjust-latency = false
|
||||
|
||||
##
|
||||
## ALSA backend stuff
|
||||
##
|
||||
[alsa]
|
||||
|
||||
## device: (global)
|
||||
# Sets the device name for the default playback device.
|
||||
#device = default
|
||||
|
||||
## device-prefix: (global)
|
||||
# Sets the prefix used by the discovered (non-default) playback devices. This
|
||||
# will be appended with "CARD=c,DEV=d", where c is the card id and d is the
|
||||
# device index for the requested device name.
|
||||
#device-prefix = plughw:
|
||||
|
||||
## device-prefix-*: (global)
|
||||
# Card- and device-specific prefixes may be used to override the device-prefix
|
||||
# option. The option may specify the card id (eg, device-prefix-NVidia), or
|
||||
# the card id and device index (eg, device-prefix-NVidia-0). The card id is
|
||||
# case-sensitive.
|
||||
#device-prefix- =
|
||||
|
||||
## custom-devices: (global)
|
||||
# Specifies a list of enumerated playback devices and the ALSA devices they
|
||||
# refer to. The list pattern is "Display Name=ALSA device;...". The display
|
||||
# names will be returned for device enumeration, and the ALSA device is the
|
||||
# device name to open for each enumerated device.
|
||||
#custom-devices =
|
||||
|
||||
## capture: (global)
|
||||
# Sets the device name for the default capture device.
|
||||
#capture = default
|
||||
|
||||
## capture-prefix: (global)
|
||||
# Sets the prefix used by the discovered (non-default) capture devices. This
|
||||
# will be appended with "CARD=c,DEV=d", where c is the card id and d is the
|
||||
# device number for the requested device name.
|
||||
#capture-prefix = plughw:
|
||||
|
||||
## capture-prefix-*: (global)
|
||||
# Card- and device-specific prefixes may be used to override the
|
||||
# capture-prefix option. The option may specify the card id (eg,
|
||||
# capture-prefix-NVidia), or the card id and device index (eg,
|
||||
# capture-prefix-NVidia-0). The card id is case-sensitive.
|
||||
#capture-prefix- =
|
||||
|
||||
## custom-captures: (global)
|
||||
# Specifies a list of enumerated capture devices and the ALSA devices they
|
||||
# refer to. The list pattern is "Display Name=ALSA device;...". The display
|
||||
# names will be returned for device enumeration, and the ALSA device is the
|
||||
# device name to open for each enumerated device.
|
||||
#custom-captures =
|
||||
|
||||
## mmap:
|
||||
# Sets whether to try using mmap mode (helps reduce latencies and CPU
|
||||
# consumption). If mmap isn't available, it will automatically fall back to
|
||||
# non-mmap mode. True, yes, on, and non-0 values will attempt to use mmap. 0
|
||||
# and anything else will force mmap off.
|
||||
#mmap = true
|
||||
|
||||
## allow-resampler:
|
||||
# Specifies whether to allow ALSA's built-in resampler. Enabling this will
|
||||
# allow the playback device to be set to a different sample rate than the
|
||||
# actual output, causing ALSA to apply its own resampling pass after OpenAL
|
||||
# Soft resamples and mixes the sources and effects for output.
|
||||
#allow-resampler = false
|
||||
|
||||
##
|
||||
## OSS backend stuff
|
||||
##
|
||||
[oss]
|
||||
|
||||
## device: (global)
|
||||
# Sets the device name for OSS output.
|
||||
#device = /dev/dsp
|
||||
|
||||
## capture: (global)
|
||||
# Sets the device name for OSS capture.
|
||||
#capture = /dev/dsp
|
||||
|
||||
##
|
||||
## Solaris backend stuff
|
||||
##
|
||||
[solaris]
|
||||
|
||||
## device: (global)
|
||||
# Sets the device name for Solaris output.
|
||||
#device = /dev/audio
|
||||
|
||||
##
|
||||
## QSA backend stuff
|
||||
##
|
||||
[qsa]
|
||||
|
||||
##
|
||||
## JACK backend stuff
|
||||
##
|
||||
[jack]
|
||||
|
||||
## spawn-server: (global)
|
||||
# Attempts to autospawn a JACK server whenever needed (initializing the
|
||||
# backend, opening devices, etc).
|
||||
#spawn-server = false
|
||||
|
||||
## custom-devices: (global)
|
||||
# Specifies a list of enumerated devices and the ports they connect to. The
|
||||
# list pattern is "Display Name=ports regex;Display Name=ports regex;...". The
|
||||
# display names will be returned for device enumeration, and the ports regex
|
||||
# is the regular expression to identify the target ports on the server (as
|
||||
# given by the jack_get_ports function) for each enumerated device.
|
||||
#custom-devices =
|
||||
|
||||
## connect-ports:
|
||||
# Attempts to automatically connect the client ports to physical server ports.
|
||||
# Client ports that fail to connect will leave the remaining channels
|
||||
# unconnected and silent (the device format won't change to accommodate).
|
||||
#connect-ports = true
|
||||
|
||||
## buffer-size:
|
||||
# Sets the update buffer size, in samples, that the backend will keep buffered
|
||||
# to handle the server's real-time processing requests. This value must be a
|
||||
# power of 2, or else it will be rounded up to the next power of 2. If it is
|
||||
# less than JACK's buffer update size, it will be clamped. This option may
|
||||
# be useful in case the server's update size is too small and doesn't give the
|
||||
# mixer time to keep enough audio available for the processing requests.
|
||||
#buffer-size = 0
|
||||
|
||||
##
|
||||
## WASAPI backend stuff
|
||||
##
|
||||
[wasapi]
|
||||
|
||||
##
|
||||
## DirectSound backend stuff
|
||||
##
|
||||
[dsound]
|
||||
|
||||
##
|
||||
## Windows Multimedia backend stuff
|
||||
##
|
||||
[winmm]
|
||||
|
||||
##
|
||||
## PortAudio backend stuff
|
||||
##
|
||||
[port]
|
||||
|
||||
## device: (global)
|
||||
# Sets the device index for output. Negative values will use the default as
|
||||
# given by PortAudio itself.
|
||||
#device = -1
|
||||
|
||||
## capture: (global)
|
||||
# Sets the device index for capture. Negative values will use the default as
|
||||
# given by PortAudio itself.
|
||||
#capture = -1
|
||||
|
||||
##
|
||||
## Wave File Writer stuff
|
||||
##
|
||||
[wave]
|
||||
|
||||
## file: (global)
|
||||
# Sets the filename of the wave file to write to. An empty name prevents the
|
||||
# backend from opening, even when explicitly requested.
|
||||
# THIS WILL OVERWRITE EXISTING FILES WITHOUT QUESTION!
|
||||
#file =
|
||||
|
||||
## bformat: (global)
|
||||
# Creates AMB format files using first-order ambisonics instead of a standard
|
||||
# single- or multi-channel .wav file.
|
||||
#bformat = false
|
||||
Binary file not shown.
|
|
@ -1,43 +0,0 @@
|
|||
# AmbDec configuration
|
||||
# Written by Ambisonic Decoder Toolbox, version 8.0
|
||||
|
||||
/description 3D7_2h1v_allrad_5200_rE_max_1_band
|
||||
|
||||
/version 3
|
||||
|
||||
/dec/chan_mask 1bf
|
||||
/dec/freq_bands 1
|
||||
/dec/speakers 7
|
||||
/dec/coeff_scale fuma
|
||||
|
||||
/opt/input_scale fuma
|
||||
/opt/nfeff_comp output
|
||||
/opt/delay_comp on
|
||||
/opt/level_comp on
|
||||
/opt/xover_freq 400.000000
|
||||
/opt/xover_ratio 0.000000
|
||||
|
||||
/speakers/{
|
||||
# id dist azim elev conn
|
||||
#-----------------------------------------------------------------------
|
||||
add_spkr LF 1.500000 51.000000 24.000000
|
||||
add_spkr RF 1.500000 -51.000000 24.000000
|
||||
add_spkr CE 1.500000 0.000000 0.000000
|
||||
add_spkr LB 1.500000 180.000000 55.000000
|
||||
add_spkr RB 1.500000 0.000000 -55.000000
|
||||
add_spkr LS 1.500000 129.000000 -24.000000
|
||||
add_spkr RS 1.500000 -129.000000 -24.000000
|
||||
/}
|
||||
|
||||
/matrix/{
|
||||
order_gain 1.000000 0.774597 0.400000 0.000000
|
||||
add_row 0.325031 0.357638 0.206500 0.234037 0.202440 0.135692 0.116927 -0.098768
|
||||
add_row 0.325036 -0.357619 0.206537 0.234033 -0.202427 -0.135680 0.116934 -0.098768
|
||||
add_row 0.080073 -0.000010 -0.000296 0.155843 -0.000016 -0.000011 -0.000623 0.163306
|
||||
add_row 0.353556 0.000002 0.408453 -0.288377 -0.000004 -0.000003 -0.221039 0.077297
|
||||
add_row 0.325297 0.000008 -0.414018 0.232789 0.000004 0.000003 -0.232940 0.018311
|
||||
add_row 0.353558 0.352704 -0.203542 -0.290124 -0.191868 -0.134582 0.110616 -0.038294
|
||||
add_row 0.353556 -0.352691 -0.203576 -0.290115 0.191871 0.134585 0.110612 -0.038293
|
||||
/}
|
||||
|
||||
/end
|
||||
|
|
@ -1,51 +0,0 @@
|
|||
# AmbDec configuration
|
||||
# Written by Ambisonic Decoder Toolbox, version 8.0
|
||||
|
||||
/description Hexagon_2h0p_pinv_match_rV_max_rE_2_band
|
||||
|
||||
/version 3
|
||||
|
||||
/dec/chan_mask 11b
|
||||
/dec/freq_bands 2
|
||||
/dec/speakers 6
|
||||
/dec/coeff_scale fuma
|
||||
|
||||
/opt/input_scale fuma
|
||||
/opt/nfeff_comp input
|
||||
/opt/delay_comp on
|
||||
/opt/level_comp on
|
||||
/opt/xover_freq 400.000000
|
||||
/opt/xover_ratio 0.000000
|
||||
|
||||
/speakers/{
|
||||
# id dist azim elev conn
|
||||
#-----------------------------------------------------------------------
|
||||
add_spkr LF 1.000000 30.000000 0.000000
|
||||
add_spkr RF 1.000000 -30.000000 0.000000
|
||||
add_spkr RS 1.000000 -90.000000 0.000000
|
||||
add_spkr RB 1.000000 -150.000000 0.000000
|
||||
add_spkr LB 1.000000 150.000000 0.000000
|
||||
add_spkr LS 1.000000 90.000000 0.000000
|
||||
/}
|
||||
|
||||
/lfmatrix/{
|
||||
order_gain 1.000000 1.000000 1.000000 0.000000
|
||||
add_row 0.235702 0.166667 0.288675 0.288675 0.166667
|
||||
add_row 0.235702 -0.166667 0.288675 -0.288675 0.166667
|
||||
add_row 0.235702 -0.333333 0.000000 -0.000000 -0.333333
|
||||
add_row 0.235702 -0.166667 -0.288675 0.288675 0.166667
|
||||
add_row 0.235702 0.166667 -0.288675 -0.288675 0.166667
|
||||
add_row 0.235702 0.333333 0.000000 -0.000000 -0.333333
|
||||
/}
|
||||
|
||||
/hfmatrix/{
|
||||
order_gain 1.414214 1.224745 0.707107 0.000000
|
||||
add_row 0.235702 0.166667 0.288675 0.288675 0.166667
|
||||
add_row 0.235702 -0.166667 0.288675 -0.288675 0.166667
|
||||
add_row 0.235702 -0.333333 0.000000 -0.000000 -0.333333
|
||||
add_row 0.235702 -0.166667 -0.288675 0.288675 0.166667
|
||||
add_row 0.235702 0.166667 -0.288675 -0.288675 0.166667
|
||||
add_row 0.235702 0.333333 0.000000 -0.000000 -0.333333
|
||||
/}
|
||||
|
||||
/end
|
||||
|
|
@ -1,46 +0,0 @@
|
|||
# AmbDec configuration
|
||||
# Written by Ambisonic Decoder Toolbox, version 8.0
|
||||
|
||||
# input channel order: WYXVU
|
||||
|
||||
/description itu50-noCenter_2h0p_allrad_5200_rE_max_1_band
|
||||
|
||||
# Although unused in this configuration, the front-center is declared here so
|
||||
# that an appropriate distance may be set (for proper delaying or attenuating
|
||||
# of dialog and such which feed it directly). It otherwise does not contribute
|
||||
# to positional sound output.
|
||||
|
||||
/version 3
|
||||
|
||||
/dec/chan_mask 11b
|
||||
/dec/freq_bands 1
|
||||
/dec/speakers 5
|
||||
/dec/coeff_scale fuma
|
||||
|
||||
/opt/input_scale fuma
|
||||
/opt/nfeff_comp input
|
||||
/opt/delay_comp on
|
||||
/opt/level_comp on
|
||||
/opt/xover_freq 400.000000
|
||||
/opt/xover_ratio 0.000000
|
||||
|
||||
/speakers/{
|
||||
# id dist azim elev conn
|
||||
#-----------------------------------------------------------------------
|
||||
add_spkr LS 1.000000 110.000000 0.000000 system:playback_3
|
||||
add_spkr LF 1.000000 30.000000 0.000000 system:playback_1
|
||||
add_spkr CE 1.000000 0.000000 0.000000 system:playback_5
|
||||
add_spkr RF 1.000000 -30.000000 0.000000 system:playback_2
|
||||
add_spkr RS 1.000000 -110.000000 0.000000 system:playback_4
|
||||
/}
|
||||
|
||||
/matrix/{
|
||||
order_gain 1.00000000e+00 8.66025404e-01 5.00000000e-01 0.000000
|
||||
add_row 4.70934222e-01 3.78169605e-01 -4.00084750e-01 -8.22264454e-02 -4.43765986e-02
|
||||
add_row 2.66639870e-01 2.55418584e-01 3.32591390e-01 2.82949132e-01 8.16816772e-02
|
||||
add_row 0.00000000e+00 0.00000000e+00 0.00000000e+00 0.00000000e+00 0.00000000e+00
|
||||
add_row 2.66634915e-01 -2.55421639e-01 3.32586482e-01 -2.82947688e-01 8.16782588e-02
|
||||
add_row 4.70935891e-01 -3.78173080e-01 -4.00080588e-01 8.22279700e-02 -4.43716394e-02
|
||||
/}
|
||||
|
||||
/end
|
||||
|
|
@ -1,48 +0,0 @@
|
|||
# AmbDec configuration
|
||||
# Written by Ambisonic Decoder Toolbox, version 8.0
|
||||
|
||||
/description itu50_2h0p_allrad_5200_rE_max_1_band
|
||||
|
||||
/version 3
|
||||
|
||||
/dec/chan_mask 11b
|
||||
/dec/freq_bands 2
|
||||
/dec/speakers 5
|
||||
/dec/coeff_scale fuma
|
||||
|
||||
/opt/input_scale fuma
|
||||
/opt/nfeff_comp output
|
||||
/opt/delay_comp on
|
||||
/opt/level_comp on
|
||||
/opt/xover_freq 400.000000
|
||||
/opt/xover_ratio 3.000000
|
||||
|
||||
/speakers/{
|
||||
# id dist azim elev conn
|
||||
#-----------------------------------------------------------------------
|
||||
add_spkr LS 1.000000 110.000000 0.000000
|
||||
add_spkr LF 1.000000 30.000000 0.000000
|
||||
add_spkr CE 1.000000 0.000000 0.000000
|
||||
add_spkr RF 1.000000 -30.000000 0.000000
|
||||
add_spkr RS 1.000000 -110.000000 0.000000
|
||||
/}
|
||||
|
||||
/lfmatrix/{
|
||||
order_gain 1.000000 1.000000 1.000000 0.000000
|
||||
add_row 0.420330 0.330200 -0.312250 0.019350 -0.027010
|
||||
add_row 0.197700 0.288820 0.287820 0.049110 0.007420
|
||||
add_row 0.058030 0.000000 0.205970 0.000000 0.050790
|
||||
add_row 0.197700 -0.288820 0.287820 -0.049110 0.007420
|
||||
add_row 0.420330 -0.330200 -0.312250 -0.019350 -0.027010
|
||||
/}
|
||||
|
||||
/hfmatrix/{
|
||||
order_gain 1.000000 0.866025 0.500000 0.000000
|
||||
add_row 0.470934 0.378170 -0.400085 -0.082226 -0.044377
|
||||
add_row 0.208954 0.257988 0.230383 0.288520 -0.025085
|
||||
add_row 0.109403 -0.000002 0.194278 -0.000003 0.200863
|
||||
add_row 0.208950 -0.257989 0.230379 -0.288516 -0.025088
|
||||
add_row 0.470936 -0.378173 -0.400081 0.082228 -0.044372
|
||||
/}
|
||||
|
||||
/end
|
||||
|
|
@ -1,42 +0,0 @@
|
|||
Ambisonic decoder configuration presets are provided here for common surround
|
||||
sound speaker layouts. The presets are prepared to work with OpenAL Soft's high
|
||||
quality decoder. By default all of the speaker distances within a preset are
|
||||
set to the same value, which results in no effect from distance compensation.
|
||||
If this doesn't match your physical speaker setup, it may be worth copying the
|
||||
preset and modifying the distance values to match (note that modifying the
|
||||
azimuth and elevation values in the presets will not have any effect; the
|
||||
specified angles do not change the decoder behavior).
|
||||
|
||||
Details of the individual presets are as follows.
|
||||
|
||||
square.ambdec
|
||||
Specifies a basic square speaker setup for Quadraphonic output, with identical
|
||||
width and depth. Front speakers are placed at +45 and -45 degrees, and back
|
||||
speakers are placed at +135 and -135 degrees.
|
||||
|
||||
rectangle.ambdec
|
||||
Specifies a narrower speaker setup for Quadraphonic output, with a little less
|
||||
width but a little more depth over a basic square setup. Front speakers are
|
||||
placed at +30 and -30 degrees, providing a bit more compatibility for existing
|
||||
stereo content, with back speakers at +150 and -150 degrees.
|
||||
|
||||
itu5.1.ambdec
|
||||
Specifies a standard ITU 5.0/5.1 setup for 5.1 Surround output. The front-
|
||||
center speaker is placed directly in front at 0 degrees, with the front-left
|
||||
and front-right at +30 and -30 degrees, and the surround speakers (side or
|
||||
back) at +110 and -110 degrees.
|
||||
|
||||
hexagon.ambdec
|
||||
Specifies a flat-front hexagonal speaker setup for 7.1 Surround output. The
|
||||
front left and right speakers are placed at +30 and -30 degrees, the side
|
||||
speakers are placed at +90 and -90 degrees, and the back speakers are placed at
|
||||
+150 and -150 degrees. Although this is for 7.1 output, no front-center speaker
|
||||
is defined for the decoder, meaning that speaker will be silent for 3D sound
|
||||
(however it may still be used with AL_SOFT_direct_channels or ALC_EXT_DEDICATED
|
||||
output). A "proper" 7.1 decoder may be provided in the future, but due to the
|
||||
nature of the speaker configuration will have trade-offs.
|
||||
|
||||
3D7.1.ambdec
|
||||
Specifies a 3D7.1 speaker setup for 7.1 Surround output. Although it's for 7.1
|
||||
output, the speakers for such a configuration need to be placed in different
|
||||
positions for proper results. Please see docs/3D7.1.txt for more information.
|
||||
|
|
@ -1,45 +0,0 @@
|
|||
# AmbDec configuration
|
||||
# Written by Ambisonic Decoder Toolbox, version 8.0
|
||||
|
||||
/description Rectangle_1h0p_pinv_match_rV_max_rE_2_band
|
||||
|
||||
/version 3
|
||||
|
||||
/dec/chan_mask b
|
||||
/dec/freq_bands 2
|
||||
/dec/speakers 4
|
||||
/dec/coeff_scale fuma
|
||||
|
||||
/opt/input_scale fuma
|
||||
/opt/nfeff_comp input
|
||||
/opt/delay_comp on
|
||||
/opt/level_comp on
|
||||
/opt/xover_freq 400.000000
|
||||
/opt/xover_ratio 0.000000
|
||||
|
||||
/speakers/{
|
||||
# id dist azim elev conn
|
||||
#-----------------------------------------------------------------------
|
||||
add_spkr LF 1.000000 30.000000 0.000000
|
||||
add_spkr RF 1.000000 -30.000000 0.000000
|
||||
add_spkr RB 1.000000 -150.000000 0.000000
|
||||
add_spkr LB 1.000000 150.000000 0.000000
|
||||
/}
|
||||
|
||||
/lfmatrix/{
|
||||
order_gain 1.000000 1.000000 0.000000 0.000000
|
||||
add_row 0.353553 0.500000 0.288675
|
||||
add_row 0.353553 -0.500000 0.288675
|
||||
add_row 0.353553 -0.500000 -0.288675
|
||||
add_row 0.353553 0.500000 -0.288675
|
||||
/}
|
||||
|
||||
/hfmatrix/{
|
||||
order_gain 1.414214 1.000000 0.000000 0.000000
|
||||
add_row 0.353553 0.500000 0.288675
|
||||
add_row 0.353553 -0.500000 0.288675
|
||||
add_row 0.353553 -0.500000 -0.288675
|
||||
add_row 0.353553 0.500000 -0.288675
|
||||
/}
|
||||
|
||||
/end
|
||||
|
|
@ -1,45 +0,0 @@
|
|||
# AmbDec configuration
|
||||
# Written by Ambisonic Decoder Toolbox, version 8.0
|
||||
|
||||
/description Square_1h0p_pinv_match_rV_max_rE_2_band
|
||||
|
||||
/version 3
|
||||
|
||||
/dec/chan_mask b
|
||||
/dec/freq_bands 2
|
||||
/dec/speakers 4
|
||||
/dec/coeff_scale fuma
|
||||
|
||||
/opt/input_scale fuma
|
||||
/opt/nfeff_comp input
|
||||
/opt/delay_comp on
|
||||
/opt/level_comp on
|
||||
/opt/xover_freq 400.000000
|
||||
/opt/xover_ratio 0.000000
|
||||
|
||||
/speakers/{
|
||||
# id dist azim elev conn
|
||||
#-----------------------------------------------------------------------
|
||||
add_spkr LF 1.000000 45.000000 0.000000
|
||||
add_spkr RF 1.000000 -45.000000 0.000000
|
||||
add_spkr RB 1.000000 -135.000000 0.000000
|
||||
add_spkr LB 1.000000 135.000000 0.000000
|
||||
/}
|
||||
|
||||
/lfmatrix/{
|
||||
order_gain 1.000000 1.000000 0.000000 0.000000
|
||||
add_row 0.353553 0.353553 0.353553
|
||||
add_row 0.353553 -0.353553 0.353553
|
||||
add_row 0.353553 -0.353553 -0.353553
|
||||
add_row 0.353553 0.353553 -0.353553
|
||||
/}
|
||||
|
||||
/hfmatrix/{
|
||||
order_gain 1.414214 1.000000 0.000000 0.000000
|
||||
add_row 0.353553 0.353553 0.353553
|
||||
add_row 0.353553 -0.353553 0.353553
|
||||
add_row 0.353553 -0.353553 -0.353553
|
||||
add_row 0.353553 0.353553 -0.353553
|
||||
/}
|
||||
|
||||
/end
|
||||
Loading…
Reference in New Issue