diff --git a/ohos/arm64-v8a/share/man/man1/mpg123.1 b/ohos/arm64-v8a/share/man/man1/mpg123.1 deleted file mode 100644 index f5a88c90..00000000 --- a/ohos/arm64-v8a/share/man/man1/mpg123.1 +++ /dev/null @@ -1,504 +0,0 @@ -.TH mpg123 1 "31 Jan 2008" -.SH NAME -mpg123 \- play audio MPEG 1.0/2.0/2.5 stream (layers 1, 2 and 3) -.SH SYNOPSIS -.B mpg123 -[ -.B options -] -.IR file " ... | " URL " ... | " -.B \- -.SH DESCRIPTION -.B mpg123 -reads one or more -.IR file\^ s -(or standard input if ``\-'' is specified) or -.IR URL\^ s -and plays them on the audio device (default) or -outputs them to stdout. -.IR file\^ / URL -is assumed to be an MPEG audio bit stream. -.SH OPERANDS -The following operands are supported: -.TP 8 -.IR file (s) -The path name(s) of one or more input files. They must be -valid MPEG-1.0/2.0/2.5 audio layer 1, 2 or 3 bit streams. -If a dash ``\-'' is specified, MPEG data will -be read from the standard input. Furthermore, any name -starting with ``http://'' is recognized as -.I URL -(see next section). -.SH OPTIONS -.B mpg123 -options may be either the traditional POSIX one letter options, -or the GNU style long options. POSIX style options start with a -single ``\-'', while GNU long options start with ``\-\^\-''. -Option arguments (if needed) follow separated by whitespace (not ``=''). -Note that some options can be absent from your installation when disabled in the build process. -.SH INPUT OPTIONS -.TP -\fB\-k \fInum\fR, \fB\-\^\-skip \fInum -Skip first -.I num -frames. By default the decoding starts at the first frame. -.TP -\fB\-n \fInum\fR, \fB\-\^\-frames \fInum -Decode only -.I num -frames. By default the complete stream is decoded. -.TP -.BR \-\-fuzzy -Enable fuzzy seeks (guessing byte offsets or using approximate seek points from Xing TOC). -Without that, seeks need a first scan through the file before they can jump at positions. -You can decide here: sample-accurate operation with gapless features or faster (fuzzy) seeking. -.TP -.BR \-y ", " \-\^\-no\-resync -Do NOT try to resync and continue decoding if an error occurs in -the input file. Normally, -.B mpg123 -tries to keep the playback alive at all costs, including skipping invalid material and searching new header when something goes wrong. -With this switch you can make it bail out on data errors -(and perhaps spare your ears a bad time). Note that this switch has been renamed from \-\-resync. -The old name still works, but is not advertised or recommened to use (subject to removal in future). -.TP -\fB\-\^-resync\-limit \fIbytes\fR -Set number of bytes to search for valid MPEG data; <0 means search whole stream. -If you know there are huge chunks of invalid data in your files... here is your hammer. -.TP -\fB\-p \fIURL \fR| \fBnone\fR, \fB\-\^\-proxy \fIURL \fR| \fBnone -The specified -.I proxy -will be used for HTTP requests. It -should be specified as full URL (``http://host.domain:port/''), -but the ``http://'' prefix, the port number and the trailing -slash are optional (the default port is 80). Specifying -.B none -means not to use any proxy, and to retrieve files directly -from the respective servers. See also the -``HTTP SUPPORT'' section. -.TP -\fB\-u \fIauth\fR, \fB\-\^\-auth \fIauth -HTTP authentication to use when recieving files via HTTP. -The format used is user:password. -.TP -\fB\-@ \fIfile\fR, \fB\-\^\-list \fIfile -Read filenames and/or URLs of MPEG audio streams from the specified -.I file -in addition to the ones specified on the command line (if any). -Note that -.I file -can be either an ordinary file, a dash ``\-'' to indicate that -a list of filenames/URLs is to be read from the standard input, -or an URL pointing to a an appropriate list file. Note: only -one -.B \-@ -option can be used (if more than one is specified, only the -last one will be recognized). -.TP -\fB\-l \fIn\fR, \fB\-\^\-listentry \fIn -Of the playlist, play specified entry only. -.I n -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... -.TP -\fB\-\-loop \fItimes\fR -for looping track(s) a certain number of times, < 0 means infinite loop (not with --random!). -.TP -.BR \-\-keep\-open -For remote control mode: Keep loaded file open after reaching end. -.TP -\fB\-\-timeout \fIseconds\fR -Timeout in (integer) seconds before declaring a stream dead (if <= 0, wait forever). -.TP -.BR \-z ", " \-\^\-shuffle -Shuffle play. Randomly shuffles the order of files specified on the command -line, or in the list file. -.TP -.BR \-Z ", " \-\-random -Continuous random play. Keeps picking a random file from the command line -or the play list. Unlike shuffle play above, random play never ends, and -plays individual songs more than once. -.TP -\fB\-\^\-no\-icy\-meta -Do not accept ICY meta data. -.TP -\fB\-i, \-\^-\index -Index / scan through the track before playback. -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. -.TP -\fB\-\-index\-size \fIsize\fR -Set the number of entries in the seek frame index table. -.TP -\fB\-\-preframes \fInum\fR -Set the number of frames to be read as lead-in before a seeked-to position. -This serves to fill the layer 3 bit reservoir, which is needed to faithfully reproduce a certain sample at a certain position. -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). - -.SH OUTPUT and PROCESSING OPTIONS -.TP -\fB\-o \fImodule\fR, \-\^\-output \fImodule\fR -Select audio output module. You can provide a comma-separated list to use the first one that works. -.TP -\fB\-\^\-list\-modules -List the available modules. -.TP -\fB\-a \fIdev\fR, \fB\-\^\-audiodevice \fIdev -Specify the audio device to use. The default is -system-dependent (usually /dev/audio or /dev/dsp). -Use this option if you have multiple audio devices and -the default is not what you want. -.TP -.BR \-s ", " \-\^\-stdout -The decoded audio samples are written to standard output, -instead of playing them through the audio device. This -option must be used if your audio hardware is not supported -by -.BR mpg123 . -The output format per default is raw (headerless) linear PCM audio data, -16 bit, stereo, host byte order (you can force mono or 8bit). -.TP -\fB\-O \fIfile\fR, \fB\-\^\-outfile -Write raw output into a file (instead of simply redirecting standard output to a file with the shell). -.TP -\fB\-w \fIfile\fR, \fB\-\^\-wav -Write output as WAV file. This will cause the MPEG stream to be decoded -and saved as file -.I file -, or standard output if -.I - -is used as file name. You can also use -.I --au -and -.I --cdr -for AU and CDR format, respectively. -.TP -\fB\-\^\-au \fIfile -Does not play the MPEG file but writes it to -.I file -in SUN audio format. If \- is used as the filename, the AU file is -written to stdout. -.TP -\fB\-\^\-cdr \fIfile -Does not play the MPEG file but writes it to -.I file -as a CDR file. If \- is used as the filename, the CDR file is written -to stdout. -.TP -.BR \-\-reopen -Forces reopen of the audiodevice after ever song -.TP -.BR \-\-cpu\ \fIdecoder\-type -Selects a certain decoder (optimized for specific CPU), for example i586 or MMX. -The list of available decoders can vary; depending on the build and what your CPU supports. -This options is only availabe when the build actually includes several optimized decoders. -.TP -.BR \-\-test\-cpu -Tests your CPU and prints a list of possible choices for \-\-cpu. -.TP -.BR \-\-list\-cpu -Lists all available decoder choices, regardless of support by your CPU. -.TP -\fB\-g \fIgain\fR, \fB\-\^\-gain \fIgain -[DEPRECATED] Set audio hardware output gain (default: don't change). The unit of the gain value is hardware and output module dependent. -(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!) -.TP -\fB\-f \fIfactor\fR, \fB\-\^\-scale \fIfactor -Change scale factor (default: 32768). -.TP -.BR \-\-rva-mix,\ \-\-rva-radio -Enable RVA (relative volume adjustment) using the values stored for ReplayGain radio mode / mix mode with all tracks roughly equal loudness. -The first valid information found in ID3V2 Tags (Comment named RVA or the RVA2 frame) or ReplayGain header in Lame/Info Tag is used. -.TP -.BR \-\-rva-album,\ \-\-rva-audiophile -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. -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. -.TP -.BR \-0 ", " \-\^\-single0 "; " \-1 ", " \-\^\-single1 -Decode only channel 0 (left) or channel 1 (right), -respectively. These options are available for -stereo MPEG streams only. -.TP -.BR \-m ", " \-\^\-mono ", " \-\^\-mix ", " \-\^\-singlemix -Mix both channels / decode mono. It takes less -CPU time than full stereo decoding. -.TP -.BR \-\-stereo -Force stereo output -.TP -\fB\-r \fIrate\fR, \fB\-\^\-rate \fIrate -Set sample rate (default: automatic). You may want to -change this if you need a constant bitrate independed of -the mpeg stream rate. mpg123 automagically converts the -rate. You should then combine this with \-\-stereo or \-\-mono. -.TP -.BR \-2 ", " \-\^\-2to1 "; " \-4 ", " \-\^\-4to1 -Performs a downsampling of ratio 2:1 (22 kHz) or 4:1 (11 kHz) -on the output stream, respectively. Saves some CPU cycles, but -at least the 4:1 ratio sounds ugly. -.TP -.BR \-\-pitch\ \fIvalue -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. -.TP -.BR \-\-8bit -Forces 8bit output -.TP -\fB\-d \fIn\fR, \fB\-\^\-doublespeed \fIn -Only play every -.IR n 'th -frame. This will cause the MPEG stream -to be played -.I n -times faster, which can be used for special -effects. Can also be combined with the -.B \-\^\-halfspeed -option to play 3 out of 4 frames etc. Don't expect great -sound quality when using this option. -.TP -\fB\-h \fIn\fR, \fB\-\^\-halfspeed \fIn -Play each frame -.I n -times. This will cause the MPEG stream -to be played at -.IR 1 / n 'th -speed (n times slower), which can be -used for special effects. Can also be combined with the -.B \-\^\-doublespeed -option to double every third frame or things like that. -Don't expect great sound quality when using this option. -.TP -\fB\-E \fIfile\fR, \fB\-\^\-equalizer -Enables equalization, taken from -.IR file . -The file needs to contain 32 lines of data, additional comment lines may -be prefixed with -.IR # . -Each data line consists of two floating-point entries, separated by -whitespace. They specify the multipliers for left and right channel of -a certain frequency band, respectively. The first line corresponds to the -lowest, the 32nd to the highest frequency band. -Note that you can control the equalizer interactively with the generic control interface. -.TP -\fB\-\^\-gapless -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. -This is enabled per default beginning with mpg123 version 1.0.0 . -.TP -\fB\-\^\-no\-gapless -Disable the gapless code. That gives you MP3 decodings that include encoder delay and padding plus mpg123's decoder delay. -.TP -\fB\-D \fIn\fR, \fB\-\-delay \fIn -Insert a delay of \fIn\fR seconds before each track. -.TP -.BR "\-o h" ", " \-\^\-headphones -Direct audio output to the headphone connector (some hardware only; AIX, HP, SUN). -.TP -.BR "\-o s" ", " \-\^\-speaker -Direct audio output to the speaker (some hardware only; AIX, HP, SUN). -.TP -.BR "\-o l" ", " \-\^\-lineout -Direct audio output to the line-out connector (some hardware only; AIX, HP, SUN). -.TP -\fB\-b \fIsize\fR, \fB\-\^\-buffer \fIsize -Use an audio output buffer of -.I size -Kbytes. This is useful to bypass short periods of heavy -system activity, which would normally cause the audio output -to be interrupted. -You should specify a buffer size of at least 1024 -(i.e. 1 Mb, which equals about 6 seconds of audio data) or more; -less than about 300 does not make much sense. The default is 0, -which turns buffering off. -.TP -\fB\-\^\-preload \fIfraction -Wait for the buffer to be filled to -.I fraction -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. -Default is 1 (wait for full buffer before playback). -.TP -\fB\-\^\-smooth -Keep buffer over track boundaries -- meaning, do not empty the buffer between tracks for possibly some added smoothness. - -.SH MISC OPTIONS - -.TP -.BR \-t ", " \-\^\-test -Test mode. The audio stream is decoded, but no output occurs. -.TP -.BR \-c ", " \-\^\-check -Check for filter range violations (clipping), and report them for each frame -if any occur. -.TP -.BR \-v ", " \-\^\-verbose -Increase the verbosity level. For example, displays the frame -numbers during decoding. -.TP -.BR \-q ", " \-\^\-quiet -Quiet. Suppress diagnostic messages. -.TP -.BR \-C ", " \-\^\-control -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 -beginning of the song, ',' to rewind, '.' to fast forward, and 'q' to quit. -Type 'h' for a full list of available controls. -.TP -\fB\-\^\-title -In an xterm, or rxvt (compatible, TERM environment variable is examined), change the window's title to the name of song currently -playing. -.TP -\fB\-\^\-long\-tag -Display ID3 tag info always in long format with one line per item (artist, title, ...) -.TP -.BR \-\-utf8 -Regardless of environment, print metadata in UTF-8 (otherwise, when not using UTF-8 locale, you'll get ASCII stripdown). -.TP -.BR \-R ", " \-\^\-remote -Activate generic control interface. -.B mpg123 -will then read and execute commands from stdin. Basic usage is ``load '' to play some file and the obvious ``pause'', ``command. -``jump '' will jump/seek to a given point (MPEG frame number). -Issue ``help'' to get a full list of commands and syntax. -.TP -.BR \-\^\-remote\-err -Print responses for generic control mode to standard error, not standard out. -This is automatically triggered when using -.B -s -\fN. -.TP -\fB\-\-fifo \fIpath -Create a fifo / named pipe on the given path and use that for reading commands instead of standard input. -.TP -\fB\-\^\-aggressive -Tries to get higher priority -.TP -.BR \-T ", " \-\-realtime -Tries to gain realtime priority. This option usually requires root -privileges to have any effect. -.TP -.BR \-? ", " \-\^\-help -Shows short usage instructions. -.TP -.BR \-\^\-longhelp -Shows long usage instructions. -.TP -.BR \-\^\-version -Print the version string. -.SH HTTP SUPPORT -In addition to reading MPEG audio streams from ordinary -files and from the standard input, -.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 -specified using a so-called URL, which starts with ``http://''. When a file with -that prefix is encountered, -.B mpg123 -attempts to open an HTTP connection to the server in order to -retrieve that file to decode and play it. -.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. -.P -Note that, in order to play MPEG audio files from a WWW -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 , -.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 diff --git a/ohos/arm64-v8a/share/openal/alsoftrc.sample b/ohos/arm64-v8a/share/openal/alsoftrc.sample deleted file mode 100644 index 3c964ada..00000000 --- a/ohos/arm64-v8a/share/openal/alsoftrc.sample +++ /dev/null @@ -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 diff --git a/ohos/arm64-v8a/share/openal/hrtf/Default HRTF.mhr b/ohos/arm64-v8a/share/openal/hrtf/Default HRTF.mhr deleted file mode 100644 index 516a6786..00000000 Binary files a/ohos/arm64-v8a/share/openal/hrtf/Default HRTF.mhr and /dev/null differ diff --git a/ohos/arm64-v8a/share/openal/presets/3D7.1.ambdec b/ohos/arm64-v8a/share/openal/presets/3D7.1.ambdec deleted file mode 100644 index 42b6a0bb..00000000 --- a/ohos/arm64-v8a/share/openal/presets/3D7.1.ambdec +++ /dev/null @@ -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 diff --git a/ohos/arm64-v8a/share/openal/presets/hexagon.ambdec b/ohos/arm64-v8a/share/openal/presets/hexagon.ambdec deleted file mode 100644 index d45f2732..00000000 --- a/ohos/arm64-v8a/share/openal/presets/hexagon.ambdec +++ /dev/null @@ -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 diff --git a/ohos/arm64-v8a/share/openal/presets/itu5.1-nocenter.ambdec b/ohos/arm64-v8a/share/openal/presets/itu5.1-nocenter.ambdec deleted file mode 100644 index 23839d0e..00000000 --- a/ohos/arm64-v8a/share/openal/presets/itu5.1-nocenter.ambdec +++ /dev/null @@ -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 diff --git a/ohos/arm64-v8a/share/openal/presets/itu5.1.ambdec b/ohos/arm64-v8a/share/openal/presets/itu5.1.ambdec deleted file mode 100644 index 74386034..00000000 --- a/ohos/arm64-v8a/share/openal/presets/itu5.1.ambdec +++ /dev/null @@ -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 diff --git a/ohos/arm64-v8a/share/openal/presets/presets.txt b/ohos/arm64-v8a/share/openal/presets/presets.txt deleted file mode 100644 index 541416e2..00000000 --- a/ohos/arm64-v8a/share/openal/presets/presets.txt +++ /dev/null @@ -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. diff --git a/ohos/arm64-v8a/share/openal/presets/rectangle.ambdec b/ohos/arm64-v8a/share/openal/presets/rectangle.ambdec deleted file mode 100644 index caf72318..00000000 --- a/ohos/arm64-v8a/share/openal/presets/rectangle.ambdec +++ /dev/null @@ -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 diff --git a/ohos/arm64-v8a/share/openal/presets/square.ambdec b/ohos/arm64-v8a/share/openal/presets/square.ambdec deleted file mode 100644 index 547ed367..00000000 --- a/ohos/arm64-v8a/share/openal/presets/square.ambdec +++ /dev/null @@ -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 diff --git a/ohos/x86_64/share/openal/alsoftrc.sample b/ohos/x86_64/share/openal/alsoftrc.sample deleted file mode 100644 index 3c964ada..00000000 --- a/ohos/x86_64/share/openal/alsoftrc.sample +++ /dev/null @@ -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 diff --git a/ohos/x86_64/share/openal/hrtf/Default HRTF.mhr b/ohos/x86_64/share/openal/hrtf/Default HRTF.mhr deleted file mode 100644 index 516a6786..00000000 Binary files a/ohos/x86_64/share/openal/hrtf/Default HRTF.mhr and /dev/null differ diff --git a/ohos/x86_64/share/openal/presets/3D7.1.ambdec b/ohos/x86_64/share/openal/presets/3D7.1.ambdec deleted file mode 100644 index 42b6a0bb..00000000 --- a/ohos/x86_64/share/openal/presets/3D7.1.ambdec +++ /dev/null @@ -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 diff --git a/ohos/x86_64/share/openal/presets/hexagon.ambdec b/ohos/x86_64/share/openal/presets/hexagon.ambdec deleted file mode 100644 index d45f2732..00000000 --- a/ohos/x86_64/share/openal/presets/hexagon.ambdec +++ /dev/null @@ -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 diff --git a/ohos/x86_64/share/openal/presets/itu5.1-nocenter.ambdec b/ohos/x86_64/share/openal/presets/itu5.1-nocenter.ambdec deleted file mode 100644 index 23839d0e..00000000 --- a/ohos/x86_64/share/openal/presets/itu5.1-nocenter.ambdec +++ /dev/null @@ -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 diff --git a/ohos/x86_64/share/openal/presets/itu5.1.ambdec b/ohos/x86_64/share/openal/presets/itu5.1.ambdec deleted file mode 100644 index 74386034..00000000 --- a/ohos/x86_64/share/openal/presets/itu5.1.ambdec +++ /dev/null @@ -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 diff --git a/ohos/x86_64/share/openal/presets/presets.txt b/ohos/x86_64/share/openal/presets/presets.txt deleted file mode 100644 index 541416e2..00000000 --- a/ohos/x86_64/share/openal/presets/presets.txt +++ /dev/null @@ -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. diff --git a/ohos/x86_64/share/openal/presets/rectangle.ambdec b/ohos/x86_64/share/openal/presets/rectangle.ambdec deleted file mode 100644 index caf72318..00000000 --- a/ohos/x86_64/share/openal/presets/rectangle.ambdec +++ /dev/null @@ -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 diff --git a/ohos/x86_64/share/openal/presets/square.ambdec b/ohos/x86_64/share/openal/presets/square.ambdec deleted file mode 100644 index 547ed367..00000000 --- a/ohos/x86_64/share/openal/presets/square.ambdec +++ /dev/null @@ -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