./0040755000000000000000000000000010032107733007711 5ustar rootrootinstall/0040755000000000000000000000000010032107733011222 5ustar rootrootinstall/doinst.sh0100644000000000000000000000035610032107733013057 0ustar rootroot( cd usr/bin ; rm -rf rec ) ( cd usr/bin ; ln -sf play rec ) ( cd usr/man/man1 ; rm -rf rec.1.gz ) ( cd usr/man/man1 ; ln -sf play.1.gz rec.1.gz ) ( cd usr/man/man1 ; rm -rf soxmix.1.gz ) ( cd usr/man/man1 ; ln -sf sox.1.gz soxmix.1.gz ) install/slack-desc0100644000000000000000000000130610032107733013153 0ustar rootroot# HOW TO EDIT THIS FILE: # The "handy ruler" below makes it easier to edit a package description. Line # up the first '|' above the ':' following the base package name, and the '|' # on the right side marks the last column you can put a character in. You must # make exactly 11 lines for the formatting to be correct. It's also # customary to leave one space after the ':'. |-----handy-ruler------------------------------------------------------| sox: sox (universal sound sample translator) sox: sox: Sox is a universal sound converter, player, and recorder. It can sox: play, record, and convert between several sound file formats such as sox: .au, .snd, .voc and .wav. sox: sox: sox: sox: sox: sox: usr/0040755000000000000000000000000010032107733010365 5ustar rootrootusr/bin/0040755000000000000010000000000010032107733010723 5ustar rootbinusr/bin/play0100755000000000000010000001222410032107147011613 0ustar rootbin#!/bin/sh # Shell script to play/record sound files to/from unix style sound devices. # Should auto detect most supported systems. # # Originally developed by Chris Bagwell (cbagwell@sprynet.com) # # TODO: Put each set of fopts and filenames on an array and then # play each filename back with the given effect.h # # Change History: # # Major updates have been supplied by Kjetil Torgrim Homme and # Kirk Goff. # Set up path so that it can find Sox if user's path doesn't already # include it. prefix=/usr exec_prefix=${prefix} bindir=${exec_prefix}/bin # Look for sox in install directory first and then in current direction # if not found. PATH="$bindir:.:$PATH" program_name=`basename $0` program_version="2.0" if [ -z "$1" ]; then echo "\ $program_name: too few arguments Try \`$program_name --help' for more information." 1>&2 exit 1 fi version() { echo "$program_name (sox) $program_version" exit 0 } help() { echo "\ Usage: $program_name [OPTION]... FILE [EFFECT]... Play/record sound files to/from unix style sound devices. -c, --channels=CHANNELS specifies the number of sound channels in FILE -d, --device=DEVICE use DEVICE for input/output -f, --format=FORMAT specifies bit format of sample FORMAT is either s, u, U, A, a, or g -r, --rate=RATE sample rate in hertz of FILE -s, --size=SIZE interpret size of sample SIZE is either b, w, l, f, d, or D -t, --type=TYPE specifies file format of FILE -v, --volume=VOLUME change amplitude -x, --xinu reverse bit order of sample (only works with 16-bit and 32-bit integer data) --file next argument is FILE -h, --help display this help and exit --version output version information and exit EFFECTs are one or more of the following: avg, band, chorus, copy, cut, deemph, echo, echos, flanger, highp, lowp, map, mask, phaser, pick, polyphase rate, resample, reverb, reverse, split, stat, vibro. See sox man page for detailed information on supported file types, data formats, and effect options." exit 0 } # loop over arguments while [ $# -ne 0 ]; do case "$1" in avg|band|bandpass|bandreject|chorus|compand|copy|cut|deemph|earwax|echo|echos|fade|filter|flanger|highp|highpass|lowp|lowpass|map|mask|pan|phaser|pick|pitch|polyphase|rate|resample|reverb|reverse|silence|speed|split|stat|stretch|swap|trim|vibro|vol) effects="$@" break ;; -c) shift fopts="$fopts -c $1" ;; --channels=*) fopts="$fopts -c `echo $1 | sed 's/.*=//'`" ;; -d) shift device="$1" ;; --device=*) device=`echo $1 | sed 's/.*=//'` ;; -f) shift fopts="$fopts -$1" ;; --format=*) fopts="$fopts -`echo $1 | sed 's/.*=//'`" ;; -r) shift fopts="$fopts -r $1" ;; --rate=*) fopts="$fopts -r `echo $1 | sed 's/.*=//'`" ;; -s) shift fopts="$fopts -$1" ;; --size=*) fopts="$fopts -`echo $1 | sed 's/.*=//'`" ;; -t) shift fopts2="$fopts -t $1" ;; --type=*) fopts2="$fopts -t `echo $1 | sed 's/.*=//'`" ;; -v) shift volume="-v $1" ;; --volume=*) volume="-v `echo $1 | sed 's/.*=//'`" ;; -x|--xinu) fopts="$fopts -x" ;; --file) shift if [ -z "$filename" ]; then filename="$1" else echo "Filename already given. Ignoring extra name: $1" 1>&2 fi ;; -h) help ;; --help) help ;; --version) version ;; -) filename="-" ;; -*) fopts="$fopts $1" ;; *) if [ -z "$filename" ]; then filename="$1" else echo "Filename already given. Ignoring extra name: $1" 1>&2 fi ;; esac shift done arch=`uname -s` case $arch in SunOS) case `uname -r` in # Solaris software can auto-detect hardware capabilities. 5.*) arch_defines="-t sunau" ;; # For SunOS default to signed words. Some hardware can only play u-law and would need # to be changed here. *) arch_defines="-t sunau -w -s" ;; esac if [ -z "$device" ]; then device="/dev/audio" fi ;; Linux|FreeBSD) arch_defines="-t ossdsp" if [ -z "$device" ]; then device="/dev/dsp" fi ;; NetBSD) arch_defines="-t sunau" if [ -z "$device" ]; then device="/dev/audio" fi ;; esac # If user sets AUDIODEV environment variable then force output device # to by that. Solaris SunRay's make use of this for sure. if [ -n "$AUDIODEV" ]; then device="$AUDIODEV" fi # If name is "rec" then record else assume user is wanting to play # a sound file. if [ "$program_name" = "rec" ]; then # Don't send data to stdout if they are reading from it. if [ "$filename" = "-" ]; then echo "Send break (control-c) to end recording" 1>&2 else echo "Send break (control-c) to end recording" fi sox $volume $arch_defines $fopts $device $fopts2 "$filename" $effects else sox $volume $fopts $fopts2 "$filename" $arch_defines $device $effects fi usr/bin/sox0100755000000000000010000103100410032107150011447 0ustar rootbinELF@&\4-4 (4@4@4@@ @@^^^CnCnl$0D40D40$@$@$ dtPKtCKtCKt/lib/ld.so.1GNU a{cNOsS >ZbyXQpd,9[L`lM.\iq:74kt)]x mo^E0wF&feTrga6W8C(3vzKAnY +#$! -2'BD5@%=" VRHP_JhG/jI<U ;u1*?*@ @<z@\+02@|t!@@>@*@t<@,uD40@<@\J6@|3@LF@T@@X9@%@<@\P@|@ T@@hS@@@<@\22@|@@Z@t@&@~@<@ @\@|@@Z@n@J@D6D6D7E@<B@\@|@Dk@H@L@(7@@<|@\2@|;@@@@M@L@ Z`D6!@ <L/@ \R%@ |N@ \X@ bV@ b@  @!0<@!<J>@!\a@!|G@!jB @!@!@!@"x@"<t@"\4A@"|F@"@"W@"S@"@# @#<Xz@#\`YD6@#|@#D5<T@#lF*<t@#@#,D7@$H@$<@$\@$|8D7 Bs@$@$`@$n{@$j.@% @%<xj@%\@ @%|@%4 @%%D7 @%Z @%P@&m@&<libm.so.6logsqrtceilfloorldexpfmodcossinpowlog10tan_Jv_RegisterClassesfrexp__gmon_start__libogg.so.0_DYNAMICogg_stream_clearogg_stream_packetinogg_page_eosogg_stream_init_finiogg_stream_pageout_GLOBAL_OFFSET_TABLE_ogg_stream_flushlibvorbis.so.0vorbis_info_clearvorbis_bitrate_flushpacketvorbis_analysis_buffervorbis_bitrate_addblockvorbis_info_initvorbis_analysis_blockoutvorbis_analysis_headeroutvorbis_analysis_wrotevorbis_analysisvorbis_analysis_initvorbis_dsp_clearvorbis_block_initvorbis_block_clearlibvorbisfile.so.3ov_infoov_open_callbacksov_commentov_clearov_readlibvorbisenc.so.2vorbis_encode_init_vbrlibmad.so.0mad_frame_initmad_stream_skipmad_frame_decodemad_synth_initmad_stream_errorstrmad_frame_finishmad_timer_zeromad_synth_framemad_timer_addmad_stream_buffermad_stream_finishmad_stream_initlibc.so.6strcpyioctl__strtod_internalstdoutvsprintfstrerrormemmove__strtol_internalfgetstmpfilesetvbuffeofmallocoptargfflushstrrchr__ctype_tolower_loccallocctimestrcatfseekoptindstdinferrorsignalstrncmpstrncpyunlinkstrcasecmprealloc_IO_getcstrtoksscanffreadgetoptftellsrandclearerr__assert_failfclosestderrfputc__ctype_b_locfwriterewind__errno_locationexit__fxstatfopen_IO_putcfileno_IO_stdin_used__libc_start_mainstrlenstrchrvfprintfraisefree_edata__bss_start_endGLIBC_2.1GLIBC_2.3GLIBC_2.0c@ iiq ii{ ii iiD6w D6, D6- D7. 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P R!X@ >tX X X0XPH9 P4"D3 #"DX4X #3*"#*U X0$2:#P4"D3D!X4!UX 3UX $!*4CJ>CJ,CJF@CJr"PX"DK '  X8C2#*!X@ T X8C2#*! T X8C2#*T X8C2#*Y W X X X0XPH 4BƬ>P$>0$!X@3X3 4B>HP$ >B$W@@J$X@A@03@D@X3 B>B ($ *>(3D$ 0$ P4>B W *X4>(3D$ 0$   4 CJPCJ,CJj@"XPD2 X X X0XPH "H #QDQZ#F!X@$``#`BPTS(8ͧ˧D/ DD*;$T VD*;"3 $T %#ͧؐ(@CnCnXP X X! X DX@𘘯,CnCnBXX P˧X! ̧X X X@𘘿0D5,XXP XtX@oktPXͧ%\X0 M X@oxCan only specify "-e" for output filenamesNot enough input files specifiedNo output file?Volume factor: %f Volume adjustment is negative. This will result in a phase change %s: Known effects: Effect '%s' is not known!%s To many effects specified. autoUnknown input file format for '%s': %sCan't open input file '%s': %sUnknown output file format for '%s': %swbCan't open output file '%s': %sCan't set write buffer+r:v:t:c:phsuUAaigbwlfdxV-A-U-a-c must be given a positive number-c must be given a number-c-d-g-i-l-r must be given a positive integer-r-s-t-u%lf-v-w-xVolume value '%s' is not a numberInput file %s: using sample rate %lu size %s, encoding %s, %d %sOutput file %s: using sample rate %lu size %s, encoding %s, %d %sOutput file: comment "%s" channelInput file %s: comment "%s" Failed reading %s: %sbad input format for file %s: %sbad output format: %scould not allocate memoryError writing: %sEffect took & gave no samples!avgresampleCan not specify multiple effects that change sample rateCan not specify multiple effects that modify channel #Volume change clipped %d samples[ gopts ] [ fopts ] ifile [ fopts ] ofile [ effect [ effopts ] ]%s: Usage: %s Failed: %s gopts: -e -h -p -v volume -V fopts: -r rate -c channels -s/-u/-U/-A/-a/-i/-g/-f -b/-w/-l/-d -x effect: effopts: depends on effect Supported file formats: %s @0@/@/@/@/@/@/@/@/@/@/@/@/@/@/@/@/@/@/@/@0,@0B@/@/@/@/@/@/@/@/@/@/@0R@0h@0~@0@/@0@0@1@1@/@/@1.@/@/@/@0@/@1D@1|@1@1@1@2@20aiffaifalausndavrcdrcvscvsddatvmsdvmsgsmhcomlalumaudmp3nulossdspprcrawsfslsmpsndtsph8svxswtxwubsoufssduluwvocvorbisoggwavwvebandbandpassbandrejectchoruscompandcopydcshiftdeemphearwaxechoechosfadefilterflangerhighphighpasslowplowpassmapmaskphaserpitchpolyphasereverbreversesilencespeedstatstretchswapsynthtrimvibrovolNONSENSE!shortsNONSENSElongslong longsunsignedsigned (2's complement)u-lawa-lawfloating pointima_adpcminversed u-lawinversed A-lawMPEG audio (layer I, III or III)Version %d.%d.%dFile not seekableoffset past eofPremature EOF while reading sample file.Error writing sample file. You are probably out of disk space.File type '%s' is not knownFiletype was not specifieddata encoding %i is invaliddata size %i is invalidsampling rate was not specified%fOSS driver only supports bytes and wordsForcing to signed linear wordUnable to set the sample size to %dSound card appears to only support %d channels. Overriding format Unable to set audio speed to %d (set to %d)Unable to sync dspUnable to allocate input/output buffer of size %dInvalid audio buffer size %dstereoCouldn't set to %smonoOSS driver doesn't like unsigned bytesForcing to signed wordsOSS driver doesn't like signed wordsForcing to unsigned bytesUnable to reset OSS driver. Possibly accessing an invalid file/deviceOSS driver only supports signed with wordsForcing to signed linearOSS driver only supports unsigned with bytesFORMHeader did not begin with magic word 'FORM'8SVX'FORM' chunk does not specify '8SVX' as typeBODYVHDRANNONAMECHANCouldn't read all of headerUnable to alloc memoryUnsupported data compressionVHDR chunk has bad sizeCan't seek channel %dCan't position channel %dCan't open channel file '%s'BODY chunk not foundInvalid sample rate8svx input file must be a file, not a pipeCan't open channel output filecan't rewind output file to rewrite 8SVX headerCan't rewind channel output file %dFile created by Sound Exchange AIFF header does not begin with magic word 'FORM'AIFFAIFCAIFF 'FORM' chunk does not specify 'AIFF' or 'AIFC' as typeCOMMSSNDMARKINSTAPPLALCHCOMTAUTH(c) AIFFstartread: ignoring '%s' chunk Ignoring INSTR chunk since no MARK found.Ignoring MARK chunk since no INSTR found.Forcing data size from %d bits to 8 bitsForcing data size from %d bits to 16 bitsForcing data size from %d bits to 32 bitsunsupported sample size in AIFF header: %dYou must specify # channels, sample rate, signed/unsigned, and 8/16 on the command line.Bogus AIFF file: no COMM section.unexpected EOF while skipping AIFF offsetAIFF header has invalid blocksize. Ignoring but expect a premature EOFAIFF: no sound data on input fileCopyright:Name:Author:Comment:Annotation:sowtNONECan not support AIFC files that contain compressed data: %sMissing SSND chunk in AIFF fileUnity MIDI Note: %d Low MIDI Note: %d High MIDI Note: %d Loop %d: start: %6d end: %6d count: %6d type: forward/backward off forward AIFF Loop markers: %-10s "%s" AIFF: Unexpected EOF in %s headerAIFF: Couldn't allocate %s headerPremature EOF on AIFF input fileIgnoring AIFF tail chunk: '%s', %d bytes long You're stripping MIDI/loop info! expanding 8-bit u-law to 16 bitscan't rewind output file to rewrite AIFF headerNon-seekable file.unsupported output encoding/size for AIFF headerEOF while reading IEEE extended numberencoding: 0x%lxSorry, DEC unsupportedFound DEC magic wordInput file %s: Sun header info: %sUnexpected EOF in Sun/NeXT header info.Unsupported encoding in Sun/NeXT header. Only U-law, signed bytes, signed words, and ADPCM are supported.Sun/NeXT header size too small.Did not detect valid Sun/NeXT/DEC magic number in header.Found Sun/NeXT magic wordFound inverted Sun/NeXT magic word. Swapping bytes.Found inverted DEC magic word. Swapping bytes.Can't rewind output file to rewrite Sun header.SOXUnsupported output encoding/size for Sun/NeXT header or .AU format not specified.Only U-law, A-law signed bytes, and signed words are supported.Defaulting to 8khz u-law @X@n@@v@v@v@v@v@v@v@v@v@v@v@v@v@v@v@v@v@v@v@@v@@@Do not understand format type: %s Detected file format type: %s FSSDHCOM.snddns.RIFFCreaSOUN2BITNISTALawOgg SAMPLE DATAtive Voice FileWAVEMAUDType AUTO can only be used for input!AVR: unsupported sample resolutionAVR: unknown headerAVR: number of channels not supportedAVR: unsupported encodingAVR: file is not seekablemsb to lsbcvsd: bit rate %dbit/s, bits from %s lsb to msbcvsd: min slope %f, max slope %f cvsd: min value %f, max value %f DVMS header checksum error, read %u, calculated %u seek failed : %sunable to read DVMS header DVMS header of source file "%s": filename "%.14s" id 0x%x state 0x%x time %s usender %u ureceiver %u length %u srate %u days %u custom1 %u custom2 %u info "%.16s" DVMS rate %dbit/s using %dbit/s deviation %d%% Length in output .DVMS header will wrong since can't seek to fix itcannot write DVMS header Can't rewind output file to rewrite DVMS header. %c ; Sample Rate %ldCannot determine sample rate.; Sample Rate %ld Can only create .dat files with one channel.Forcing output to 1 channel.%*s %lgUnable to read sample. %15.8g %15.8g unable to create GSM streamgsm: channels(%d) must be in 1-16error during GSM decodewrite errorMac header type is not FSSDMac data fork is not HCOMHCOM data using value compressioncan't malloc memory for Huffman dictionaryBad sampling rate divisor in HCOM headerBad compression type in HCOM headerunexpected EOF in Mac headerunexpected EOF in HCOM dataunexpected EOF at start of HCOM datachecksum error in HCOM datanot all HCOM data readunacceptable output rate for HCOM: try 5512, 7350, 11025 or 22050 hertzcan't malloc buffer for uncompressed HCOM datacan't realloc buffer for uncompressed HCOM data Original size: %6d bytesCompressed size: %6d bytescan't malloc buffer for compressed HCOM datacan't write compressed HCOM datawrite error in HCOM headerAMAUD: header does not begin with magic word 'FORM'MAUD: 'FORM' chunk does not specify 'MAUD' as typeMDATMHDRMAUD: Unexpected EOF in ANNO headerCouldn't alloc resourcesMAUD: unsupported compression type detectedMAUD: unsupported number of channels in fileMAUD: frequency denominator == 0, failedMAUD: MHDR chunk has bad sizeMAUD: MDAT chunk not foundMAUD: unsupported number of channels, unable to storeOutput .maud file must be a file, not a pipecan't rewind output file to rewrite MAUD headerfile create by Sound eXchange @@@@@@@@@ަ@ްThe file is not an MP3 file or it is corruptedCannot determine number of channelsend of input streamread error on bitstreamCould not allocate memoryunrecoverable frame level error (%s). Sorry, no MP3 encoding supportnul fileFound Psion record.app headerFound length=%dPRC must only supports 1 channel. OverridingPRC must use 8000 sample rate. OverridingPsion header doesn't start with the correct bytes Try the '.al' (A-law raw) file type with '-t al -r 8000 filename'length now = %dHeader will be have invalid file length since file is not seekableCan't rewind output file to rewrite Psion header.Final length=%dCan't seek this data sizeUnable to alloc resourcesDo not support this encoding for this data sizeDo not support this data size for this handler@@@@@@@@@@@>@@>@>@>@H@@@@@@@@@@@@@p@0@p@p@p@z@@@@@@@@@@IRCAM comment: %sSoundfile input: unknown format 0x%x SF %s file: can't read, it is byte-swapped or it is not an IRCAM SoundFileunexpected EOF in SF headerSOUND SAMPLE DATA 2.1 EOF in SMP %.*s: %.*sSampleVision file name and comments: %sSampleVision trailer: Loop %d: start: %6ld end: %6ldMIDI Note number: %d type: forward/backward type: off type: forward SMP unable to seek back to start of sample dataunexpected EOF in SMP trailerSMP unable to seek to trailerSMP header is not version %s SMP header does not begin with magic word %s unexpected EOF in SMP headerSMP input file must be a file, not a pipe%-*sConverted using Sox.%-*.*sSMP: Can't write header completelyOutput .smp file must be a file, not a pipeSMP unable to seek back to save sizeSND: sample rate out of rangeSOUNDSND: unexpected EOF in SND headerSND: unrecognized SND formatSND: unexpected EOFcan't rewind output file to rewrite SND header%.62s - File created by Sound ExchangeNIST_1ASphere header does not begin with magic mord 'NIST_1A'end_headsample_codingsample_rate sample_byte_formatError reading Sphere header%s %s %s01%s %s %ldSPHERE format only supports ulaw and PCM data.File must be seekable for sphere file outputNIST_1A 1024 sample_count -i %ld sample_n_bytes -i %d channel_count -i %d sample_byte_format -s2 %s sample_rate -i %ld sample_coding -s3 pcm end_head sample_coding -s4 ulaw Could not rewird output file to rewrite sphere header. RRFound header filetype %sLM8953Invalid sample rate identifier found %dSample rate = %ldInvalid filetype ID in input file header, != LM8953txw input file must be a file, not a pipetx16w selected outputtx16w is overriding output format to 1 channel.tx16w is overriding output format to size Signed Word format.Output .txw file must be a file, not a pipetx16w:output finishedSound too large for TX16W. Truncating, Loop Off Creative Voice FileUnsupported VOC format CRLADPCM3 %dUnknown VOC format %dUnsupported VOC format CRLADPCM4 %dUnsupported VOC format CRLADPCM4A %dUnsupported VOC format CRLADPCM2 %dInput .voc file had no sound!VOC file header incorrectunexpected EOF in VOC headerVOC input: short fileOutput .voc file must be a file, not a pipeFile %s: only interpret 8-bit data!File %s: sample rate codes differ: %d != %dFile %s: Sample rate is zero?File %s: Silence sample rate is zeroFile %s: skipping repeat loopDon't understand size %dFile %s: skipping unknown block code %dA&A'A'A(fA(zA(A(A)A)@A-A-@Input not an Ogg Vorbis audio streamWarning: hole in stream; probably harmless Channels: %d Rate: %ld Error writing headre for Ogg Vorbis audio streamError setting up Ogg Vorbis encorder - make sure you've specied a sane rate and number of channelsbase iCoefs differ in %d/14 positionswave header error: wExtSize(%d) too small for nCoefs(%d)ADPCM file nCoefs (%.4hx) makes no sense format[%s]: samplesPerBlock(%d) incompatible with blockAlign(%d)Can only handle 4-bit MS ADPCM in wav filesformat[%s]: expects wExtSize >= %dformat[%s]: expects samplesPerBlock(%d) = %dformat[%s]: expects blockAlign(%d) = %dCan only handle 4-bit IMA ADPCM in wav fileswave header error: wExtSize inconsistent with wFmtLenwave header missing FmtExt chunkUser options overriding encoding read in .wav headerSorry, this WAV file is in IEEE Float format.Sorry, this WAV file is in OKI ADPCM format.Sorry, this WAV file is in Digistd format.Sorry, this WAV file is in Digifix format.Sorry, this WAV file is in Dolby AC2 format.Sorry, this WAV file is in Rockwell ADPCM format.Sorry, this WAV file is in Rockwell DIGITALK format.Sorry, this WAV file is in G.721 ADPCM format.Sorry, this WAV file is in G.728 CELP format.Sorry, this WAV file is in MPEG format.Sorry, this WAV file is in MPEG Layer 3 format.WAV file has unknown format type of %xSorry, this WAV file is in G.726 ADPCM format.Sorry, this WAV file is in G.722 ADPCM format.WAVE file fmt chunk is too shortPremature EOF on .wav input fileerror allocating memory for gsm buffererror setting gsm_option for WAV49 format. Recompile gsm library with -DWAV49 option and relink soxcannot create GSM objecterror during gsm decodeinvalid wav gsm frame size: %d bytesChunk %sWAVE file has missing %s chunkWAVE: RIFF header not foundWAVE header not foundfmt WAVE file is in unsupported Microsoft Official Unknown format.User options overriding channels read in .wav headerUser options overriding rate read in .wav headerSorry, don't understand .wav sizeUser options overriding size read in .wav headerReading Wave file: %s format, %d channel%s, %d samp/sec %d byte/sec, %d block align, %d bits/samp, %u data bytesLISTINFOICRDISFTcue noteadtlltxtlabl %d Extsize, %d Samps/block, %d bytes/block %d Num CoefsGSM .wav: %d Extsize, %d Samps/block, %d samples %d Extsize, %d Samps/block, %d bytes/blockLength in output .wav header will be wrong since can't seek to fix itOverriding output size to bytes for compressed data.Do not support %s in WAV files. Forcing to Signed Words.factWriting Wave file: %s format, %d channel%s, %d samp/sec %d byte/sec, %d block align, %d bits/sampFinished writing Wave file, %u data bytes %u samples GSM6.10 format: %u blocks %u padded samples %u padded data bytes help ! internal inconsistency - data_written %u gsmbytecount %uChannels(%d) must be <= 16 Overriding GSM audio from %d channel to 1 Do not support %s with 8-bit data. Forcing to unsignedDo not support %s with 16-bit data. Forcing to Signed.Can't rewind output file to rewrite .wav header.Microsoft Official UnknownMicrosoft PCMMicrosoft ADPCMIEEE FloatMicrosoft A-lawMicrosoft U-lawOKI ADPCM format.IMA ADPCMDigistd format.Digifix format.Dolby AC2GSM 6.10Rockwell ADPCMRockwell DIGITALKG.721 ADPCMG.728 CELPMPEGMPEG Layer 3UnknownG.726 ADPCMG.722 ADPCMOnly PCM SupportedAAtAA~AJAKAKAKAKAK AKAKAKAKAKAKAKAKAK0AK:AKAKAKAKJAKTAKAKAKAKAKAKAKAKAKAKAKAKAKAKAKAKAKAKAKAKAKAKAKAKAKAK^AKhAKAKAKAKAKAKAKAKAKAKxAKAKAKAKAKAKAKAKAKAKAKAKAKAKAKAKAKAKAKAKAKAKAKAKAKAKAKAKAKAKAKAKAKAKAKAKAKAKAKAKAKAKAQAQAUAUAQAUAV,AVzAZ>AZHAZPAZXAZAZAZ`AZhAZAZAZAZAZAZAZAZAZpAZxAZAZAZAZAZAZAZAZAZAZAZAZAZAZAZAZAZAZAZAZAZAZAZAZAZAZAZAZAZAZAZAZAZAZAZAZAZAZAZAZAZAZAZAZAZAZAZAZAZAZAZAZAZAZAZAZAZAZAZAZAZAZAZAZAZAZAZAZAZAZAZAZAZAZAZAZAZAZAZAZAZAZAZAZALawSoundFile**Found Psion magic wordFound PSION magic wordWVE must only supports 1 channel. OverridingWVE must use 8000 sample rate. OverridingWrong version in Psion headerFound inverted PSION magic word. Swapping bytes.Psion header doesn't start with magic word Try the '.al' file type with '-t al -r 8000 filename'avg can only take up to 16 pan valuesUsage: avg [ -l | -r | -f | -b | n,n,n...,n ]Can't average %d channels into %d channelsYou must specify at least one mix level when using avg with an unusual number of channels.Output must have different number of channels to use avg effectInput must have at least two channels to use avg -lInput must have at least two channels to use avg -rInput must have at least four channels to use avg -fInput must have at least four channels to use avg -bOutput must have known number of channels to use avg effectAbAddAdAdAdUsage: band [ -n ] center [ width ]Band: center must be < minimum data rate/2 bandpass: illegal bandwidthbandpass: illegal frequencyUsage: bandpass FREQUENCY BANDWIDTHbandreject: illegal bandwidthbandreject: illegal frequencyUsage: bandreject FREQUENCY BANDWIDTHUsage: chorus gain-in gain-out delay decay speed depth [ -s | -t ]chorus: to many delays, use less than %i delaysUsage: chorus gain-in gain-out delay decay speed [ -s | -t ]chorus: warning >>> gain-out can cause saturation or clipping of output <<<chorus: decay must be less that 1.0! chorus: decay must be positive! chorus: delay must be less than 10.0 msec! chorus: delay must be more positive! chorus: speed must be less than 5.0 Hz! chorus: speed must be more than 0.1 Hz! chorus: delay must be less than 100.0 msec! chorus: delay must be more than 20.0 msec! chorus: gain-out must be positive! chorus: gain-in must be less than 1.0! chorus: gain-in must be positive! chorus: Cannot malloc %d bytes! Out of memory,-infInput signals of zero level must always generate zero outputcompander: Odd number of transfer function parameters Each input value in dB must have a corresponding output valuecompander: Odd number of attack & decay rate parametersWrong number of arguments for the compander effect Use: {,}+ {,}+ [ [ [>> gain-out can cause saturation of output <<<echo: decay must be less than 1.0! echo: decay must be positive! echo: delay must be less than %g seconds! echo: delay must be positive! echo: gain-in must be positive! echo: gain-in must be less than 1.0! echo: Cannot malloc %d bytes! Usage: echos gain-in gain-out delay decay [ delay decay ... ]echos: to many delays, use less than %i delaysechos: warning >>> gain-out can cause saturation of output <<<echos: decay must be less than 1.0! echos: decay must be positive! echos: delay must be less than %g seconds! echos: delay must be positive! echos: gain-in must be positive! echos: gain-in must be less than 1.0! echos: Cannot malloc %d bytes! %1[qhltp]Usage: fade [ type ] fade-in-length [ stop-time [ fade-out-length ] ] Times in seconds. Fade type one of q, h, t, l or p. Fade: End of fade-in should not happen before beginning of fade-outFade: warning: End time passed end-of-file. Padding with silenceAAAAAAAAAAAAAfilter opts: %d-%d, window-len %d, beta %f Usage: filter low-high [ windowlength [ beta ] ]Usage: filter low-high [ windowlength ]filter: window length (%ld) <4 is too shortfilter: low(%d),high(%d) parameters must satisfy 0 <= low <= high <= %dfilter: adjusted freq %d-%d is identityfilter: Unable to make high filter filter: Unable to make low filter drain overran obuf by %d Usage: flanger gain-in gain-out delay decay speed [ -s | -t ]flanger: Cannot malloc %d bytes! flanger: warning >>> gain-out can cause saturation or clipping of output <<<flanger: decay must be less that 1.0! flanger: decay must be positive! flanger: speed must be less than 2.0 Hz! flanger: speed must be more than 0.1 Hz! flanger: delay must be less than 5.0 msec! flanger: delay must be positive! flanger: gain-out must be positive! flanger: gain-in must be less than 1.0! flanger: gain-in must be positive! Usage: highp cutoffHighpass: cutoff must be < sample rate / 2 (Nyquest rate) highpass: illegal frequencyUsage: highpass FREQUENCYUsage: lowp cutoffLowpass: cutoff must be < sample rate / 2 (Nyquest rate) lowpass: illegal frequencyUsage: lowpass FREQUENCYMap effect takes no options.Loop info: Loop %d: start: %6d length: %6dMIDI note: %d MIDI low : %d MIDI hi : %d Mask effect takes no options.Usage: pan direction (in [-1.0 .. 1.0])PAN onto a mono channel...PAN cannot handle different rates (in=%ld, out=%ld) use resample or rateunexpected number of channels (in=%d, out=%d)PAN clipped %d values, maybe adjust volume?Usage: phaser gain-in gain-out delay decay speed [ -s | -t ]phaser: Cannot malloc %d bytes! phaser: warning >>> gain-out can cause saturation or clipping of output <<<phaser: warning >>> gain-in can cause saturation or clipping of output <<<phaser: decay must be less that 1.0! phaser: decay must be positive! phaser: speed must be less than 2.0 Hz! phaser: speed must be more than 0.1 Hz! phaser: delay must be less than 5.0 msec! phaser: delay must be positive! Usage: pitch shift width interpole fade (in cents, in ms, cub/lin, cos/ham/lin/trap) (defaults: 0 20 c c)malloc failed in st_pitch_startPITCH cannot handle different channels (in=%ld, out=%ld) use avg or panPITCH cannot handle different rates (in=%ld, out=%ld) use resample or rateunexpected PITCH_FADE parameter %dPITCH clipped %d values... adjust volume with -v option maybe?AADADADADAADADADǍADADADADADADADAADADADADADADADADADADADADADADAADADADADAADADADǍADADADADADADADA-width-cutofflongPolyphase: unknown argument (%s %s)!factors(%d) =Number %d too large of a prime. Illegal buffer %p or length %d to nuttall. Illegal buffer %p or length %d to hamming. Illegal buffer %p, length %d, or cutoff %f. Poly: input rate %d, output rate %d. %d stages.nutPoly: window: %s size: %d cutoff: %f.Poly: stage %d: Up by %d, down by %d, i_samps %d, hsize %dPoly: : filt_len %d, cutoff freq %.1fPoly: output samples %d, oskip %dhamInput and Output rate must not be the same to use polyphase effect %)+/5;=CGIOSYaegkmq %379=KQ[]agou{  #-39;AKQWY_eikw)+57;=GUY[_mqswRate effect takes no options.rate effect can only handle rates <= 65535rate effect reduces data to 16 bitsInput and Output rates must be different to use rate effectresample: rolloff factor (%f) no good, should be 0.01>> gain-out can cause saturation of output <<<reverb: gain-out must be positivereverb: delay must be positive! reverb: Cannot malloc %d bytes! Reverse effect takes no options.Reverse effect can't create temporary file Reverse effect write error on temporary file Internal error: reverse_flow called in wrong phaseReverse effect read error from temporary file Reverse effect finds odd temporary file %lf%csilence threshold should be less than 0.0 dBsilence threshold should be between 0.0 and 100.0 %%Invalid unit specifiedUsage: silence above_periods [ duration thershold[d | %% ] ] [ below_periods duration threshold[ d | %% ]]Periods must not be greater then zeroPeriods must not be negativeUnable to allocate memoryB BdB BlBtn = %d cent = %d speed = %f speed [-c] factor (default 1.0, <1 slows, -c: factor in cent)malloc failedSPEED: %d values clipped...-s option: invalid argument-rms-freqSummary effect: unknown optionFFT can only use sample buffers of 2^n. Buffer size used is %ld Unable to allocate memory for FFT buffers. %08lx %f %f Samples read: %12u Length (seconds): %12.6f Scaled by rms: %12.6f Maximum amplitude: %12.6f Minimum amplitude: %12.6f Midline amplitude: %12.6f Mean norm: %12.6f Mean amplitude: %12.6f RMS amplitude: %12.6f Maximum delta: %12.6f Minimum delta: %12.6f Mean delta: %12.6f RMS delta: %12.6f Rough frequency: %12d Try: -t raw -b -u Try: -t raw -b -s Try: -t raw -b -U Can't guess the type Probably text, not sound Volume adjustment: %12.3f Scaled by: %12.1f %.3f Usage: stretch factor [window fade shift fading] (expansion, frame in ms, lin/..., unit<1.0, unit<0.5) (defaults: 1.0 20 lin ...) error with shift ratio valueUsage: stretch factor [window fade shift fading] (expansion, frame in ms, lin/..., unit<1.0, unit<0.5) (defaults: 1.0 20 lin ...) error with fading ratio valueUsage: stretch factor [window fade shift fading] (expansion, frame in ms, lin/..., unit<1.0, unit<0.5) (defaults: 1.0 20 lin ...) error while parsing fading ratioUsage: stretch factor [window fade shift fading] (expansion, frame in ms, lin/..., unit<1.0, unit<0.5) (defaults: 1.0 20 lin ...) error while parsing shift ratioUsage: stretch factor [window fade shift fading] (expansion, frame in ms, lin/..., unit<1.0, unit<0.5) (defaults: 1.0 20 lin ...) error while parsing fade typeUsage: stretch factor [window fade shift fading] (expansion, frame in ms, lin/..., unit<1.0, unit<0.5) (defaults: 1.0 20 lin ...) error while parsing window sizeUsage: stretch factor [window fade shift fading] (expansion, frame in ms, lin/..., unit<1.0, unit<0.5) (defaults: 1.0 20 lin ...) error while parsing factorsome malloc failedSTRETCH cannot handle different rates (in=%ld, out=%ld) use resample or rateSTRETCH cannot handle different channels (in=%d, out=%d) use avg or panSTRETCH clipped %d values...Usage: swap [1 2 | 1 2 3 4]invalid swap channel options usedCan't swap channels on mono data.sinecreatesynth: parameter errorUsage:synth [length] type mix [freq[-freq2]] [off] [ph] [p1] [p2] [p3] length in sec or hh:mm:ss.frac, 0=inputlength, default=0 is sine, square, triangle, sawtooth, trapetz, exp, whitenoise, pinknoise, brownnoise, default=sine is create, mix, amod, default=create frequency at beginning in Hz, not used for noise.. frequency at end in Hz, not used for noise.. can be given as %%n, where 'n' is the number of half notes in respect to A (440Hz) Bias (DC-offset) of signal in percent, default=0 phase shift 0..100 shift phase 0..2*Pi, not used for noise.. square: Ton/Toff, triangle+trapetz: rising slope time (0..100) trapetz: ON time (0..100) trapetz: falling slope position (0..100)synth: too many parameterssynth: illegal freq2synth: illegal freqmixamodfmodsquaresawtoothtriangleexptrapetzwhitenoisepinknoisebrownnoisevoicenoisesynth: no type givenCould not allocate memeorysynth: internel error 2synth: internal error 1synth: can not operate with more than %d channelsB=B<B=B<B=(B=B=B?FB@8B@^B@nB@BABABA>BABATrim usage: trim start [length]Usage: vibro speed [ depth ]Vibro: speed must be < 30.0, 0.0 < depth < 1.0Vibro: Cannot malloc %d bytesUsage: vol gain [ type [ limitergain ] ] (default type=amplitude: 1.0 is constant, <0.0 change phase; type=power 1.0 is constant; type=dB: 0.0 is constant, +6 doubles ampl.) The peak limiter has a gain much less than 1.0 (ie 0.05 or 0.02) which is only used on peaks to prevent clipping. 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P R!X@ >tX X X0XPH9 P4"D3 #"DX4X #3*"#*U X0$2:#P4"D3D!X4!UX 3UX $!*4CLCLpCL@CL"PX"DK '  X8C2#*!X@ T X8C2#*! T X8C2#*T X8C2#*Y W X X X0XPH 4BȘ>P$>0$!X@3X3 4B>HP$ >B$W@@J$X@A@03@D@X3 B>B ($ *>(3D$ 0$ P4>B W *X4>(3D$ 0$   4 CLCLpCL@"XPD2 X X X0XPH "H #QDQZ#F!X@$``#`BPTS(8ͧ˧D/ DD*;$T VD*;"3 $T %#ͧؐ(@CpCpXP X X! X DX@𘘯,CpCpBˬXX P˧X! ̧X X X@𘘿0D7XXP XtX@okPXͧ%ZX0 M X@oxCan only specify "-e" for output filenamesNot enough input files specifiedNo output file?Volume factor: %f Volume adjustment is negative. This will result in a phase change %s: Known effects: Effect '%s' is not known!%s To many effects specified. autoUnknown input file format for '%s': %sCan't open input file '%s': %sUnknown output file format for '%s': %swbCan't open output file '%s': %sCan't set write buffer+r:v:t:c:phsuUAaigbwlfdxV-A-U-a-c must be given a positive number-c must be given a number-c-d-g-i-l-r must be given a positive integer-r-s-t-u%lf-v-w-xVolume value '%s' is not a numberInput file %s: using sample rate %lu size %s, encoding %s, %d %sOutput file %s: using sample rate %lu size %s, encoding %s, %d %scould not allocate memoryOutput file: comment "%s" channelInput files must have the same rate, channels, data size, and encodingInput file %s: comment "%s" bad input format for file %s: %sFailed reading %s: %sbad output format: %sError writing: %sEffect took & gave no samples!avgresampleCan not specify multiple effects that change sample rateCan not specify multiple effects that modify channel #Volume change clipped %d samples[ gopts ] [ fopts ] ifile1 [fopts] ifile2 [ fopts ] ofile [ effect [ effopts ] ]%s: Usage: %s Failed: %s gopts: -e -h -p -v volume -V fopts: -r rate -c channels -s/-u/-U/-A/-a/-i/-g/-f -b/-w/-l/-d -x effect: effopts: depends on effect Supported file formats: %s @0(@/@/@/@/@/@/@/@/@/@/@/@/@/@/@/@/@/@/@/@0@@0V@/@/@/@/@/@/@/@/@/@/@0f@0|@0@0@/@0@1@1$@1,@/@/@1B@/@/@/@0@/@1X@1@1@1@1@2.@2Daiffaifalausndavrcdrcvscvsddatvmsdvmsgsmhcomlalumaudmp3nulossdspprcrawsfslsmpsndtsph8svxswtxwubsoufssduluwvocvorbisoggwavwvebandbandpassbandrejectchoruscompandcopydcshiftdeemphearwaxechoechosfadefilterflangerhighphighpasslowplowpassmapmaskphaserpitchpolyphasereverbreversesilencespeedstatstretchswapsynthtrimvibrovolNONSENSE!shortsNONSENSElongslong longsunsignedsigned (2's complement)u-lawa-lawfloating pointima_adpcminversed u-lawinversed A-lawMPEG audio (layer I, III or III)Version %d.%d.%dFile not seekableoffset past eofPremature EOF while reading sample file.Error writing sample file. You are probably out of disk space.File type '%s' is not knownFiletype was not specifieddata encoding %i is invaliddata size %i is invalidsampling rate was not specified%fOSS driver only supports bytes and wordsForcing to signed linear wordUnable to set the sample size to %dSound card appears to only support %d channels. Overriding format Unable to set audio speed to %d (set to %d)Unable to sync dspUnable to allocate input/output buffer of size %dInvalid audio buffer size %dstereoCouldn't set to %smonoOSS driver doesn't like unsigned bytesForcing to signed wordsOSS driver doesn't like signed wordsForcing to unsigned bytesUnable to reset OSS driver. Possibly accessing an invalid file/deviceOSS driver only supports signed with wordsForcing to signed linearOSS driver only supports unsigned with bytesFORMHeader did not begin with magic word 'FORM'8SVX'FORM' chunk does not specify '8SVX' as typeBODYVHDRANNONAMECHANCouldn't read all of headerUnable to alloc memoryUnsupported data compressionVHDR chunk has bad sizeCan't seek channel %dCan't position channel %dCan't open channel file '%s'BODY chunk not foundInvalid sample rate8svx input file must be a file, not a pipeCan't open channel output filecan't rewind output file to rewrite 8SVX headerCan't rewind channel output file %dFile created by Sound Exchange AIFF header does not begin with magic word 'FORM'AIFFAIFCAIFF 'FORM' chunk does not specify 'AIFF' or 'AIFC' as typeCOMMSSNDMARKINSTAPPLALCHCOMTAUTH(c) AIFFstartread: ignoring '%s' chunk Ignoring INSTR chunk since no MARK found.Ignoring MARK chunk since no INSTR found.Forcing data size from %d bits to 8 bitsForcing data size from %d bits to 16 bitsForcing data size from %d bits to 32 bitsunsupported sample size in AIFF header: %dYou must specify # channels, sample rate, signed/unsigned, and 8/16 on the command line.Bogus AIFF file: no COMM section.unexpected EOF while skipping AIFF offsetAIFF header has invalid blocksize. Ignoring but expect a premature EOFAIFF: no sound data on input fileCopyright:Name:Author:Comment:Annotation:sowtNONECan not support AIFC files that contain compressed data: %sMissing SSND chunk in AIFF fileUnity MIDI Note: %d Low MIDI Note: %d High MIDI Note: %d Loop %d: start: %6d end: %6d count: %6d type: forward/backward off forward AIFF Loop markers: %-10s "%s" AIFF: Unexpected EOF in %s headerAIFF: Couldn't allocate %s headerPremature EOF on AIFF input fileIgnoring AIFF tail chunk: '%s', %d bytes long You're stripping MIDI/loop info! expanding 8-bit u-law to 16 bitscan't rewind output file to rewrite AIFF headerNon-seekable file.unsupported output encoding/size for AIFF headerEOF while reading IEEE extended numberencoding: 0x%lxSorry, DEC unsupportedFound DEC magic wordInput file %s: Sun header info: %sUnexpected EOF in Sun/NeXT header info.Unsupported encoding in Sun/NeXT header. Only U-law, signed bytes, signed words, and ADPCM are supported.Sun/NeXT header size too small.Did not detect valid Sun/NeXT/DEC magic number in header.Found Sun/NeXT magic wordFound inverted Sun/NeXT magic word. Swapping bytes.Found inverted DEC magic word. Swapping bytes.Can't rewind output file to rewrite Sun header.SOXUnsupported output encoding/size for Sun/NeXT header or .AU format not specified.Only U-law, A-law signed bytes, and signed words are supported.Defaulting to 8khz u-law @D@Z@t@b@b@b@b@b@b@b@b@b@b@b@b@b@b@b@b@b@b@b@t@b@t@t@Do not understand format type: %s Detected file format type: %s FSSDHCOM.snddns.RIFFCreaSOUN2BITNISTALawOgg SAMPLE DATAtive Voice FileWAVEMAUDType AUTO can only be used for input!AVR: unsupported sample resolutionAVR: unknown headerAVR: number of channels not supportedAVR: unsupported encodingAVR: file is not seekablemsb to lsbcvsd: bit rate %dbit/s, bits from %s lsb to msbcvsd: min slope %f, max slope %f cvsd: min value %f, max value %f DVMS header checksum error, read %u, calculated %u seek failed : %sunable to read DVMS header DVMS header of source file "%s": filename "%.14s" id 0x%x state 0x%x time %s usender %u ureceiver %u length %u srate %u days %u custom1 %u custom2 %u info "%.16s" DVMS rate %dbit/s using %dbit/s deviation %d%% Length in output .DVMS header will wrong since can't seek to fix itcannot write DVMS header Can't rewind output file to rewrite DVMS header. %c ; Sample Rate %ldCannot determine sample rate.; Sample Rate %ld Can only create .dat files with one channel.Forcing output to 1 channel.%*s %lgUnable to read sample. %15.8g %15.8g unable to create GSM streamgsm: channels(%d) must be in 1-16error during GSM decodewrite errorMac header type is not FSSDMac data fork is not HCOMHCOM data using value compressioncan't malloc memory for Huffman dictionaryBad sampling rate divisor in HCOM headerBad compression type in HCOM headerunexpected EOF in Mac headerunexpected EOF in HCOM dataunexpected EOF at start of HCOM datachecksum error in HCOM datanot all HCOM data readunacceptable output rate for HCOM: try 5512, 7350, 11025 or 22050 hertzcan't malloc buffer for uncompressed HCOM datacan't realloc buffer for uncompressed HCOM data Original size: %6d bytesCompressed size: %6d bytescan't malloc buffer for compressed HCOM datacan't write compressed HCOM datawrite error in HCOM headerAMAUD: header does not begin with magic word 'FORM'MAUD: 'FORM' chunk does not specify 'MAUD' as typeMDATMHDRMAUD: Unexpected EOF in ANNO headerCouldn't alloc resourcesMAUD: unsupported compression type detectedMAUD: unsupported number of channels in fileMAUD: frequency denominator == 0, failedMAUD: MHDR chunk has bad sizeMAUD: MDAT chunk not foundMAUD: unsupported number of channels, unable to storeOutput .maud file must be a file, not a pipecan't rewind output file to rewrite MAUD headerfile create by Sound eXchange @@@@@,@,@@The file is not an MP3 file or it is corruptedCannot determine number of channelsend of input streamread error on bitstreamCould not allocate memoryunrecoverable frame level error (%s). Sorry, no MP3 encoding supportnul fileFound Psion record.app headerFound length=%dPRC must only supports 1 channel. OverridingPRC must use 8000 sample rate. OverridingPsion header doesn't start with the correct bytes Try the '.al' (A-law raw) file type with '-t al -r 8000 filename'length now = %dHeader will be have invalid file length since file is not seekableCan't rewind output file to rewrite Psion header.Final length=%dCan't seek this data sizeUnable to alloc resourcesDo not support this encoding for this data sizeDo not support this data size for this handler@@@@@@@@@@@*@@*@*@*@4@@@@@@@@@@@@@\@@\@\@\@f@@@@@@@@@@IRCAM comment: %sSoundfile input: unknown format 0x%x SF %s file: can't read, it is byte-swapped or it is not an IRCAM SoundFileunexpected EOF in SF headerSOUND SAMPLE DATA 2.1 EOF in SMP %.*s: %.*sSampleVision file name and comments: %sSampleVision trailer: Loop %d: start: %6ld end: %6ldMIDI Note number: %d type: forward/backward type: off type: forward SMP unable to seek back to start of sample dataunexpected EOF in SMP trailerSMP unable to seek to trailerSMP header is not version %s SMP header does not begin with magic word %s unexpected EOF in SMP headerSMP input file must be a file, not a pipe%-*sConverted using Sox.%-*.*sSMP: Can't write header completelyOutput .smp file must be a file, not a pipeSMP unable to seek back to save sizeSND: sample rate out of rangeSOUNDSND: unexpected EOF in SND headerSND: unrecognized SND formatSND: unexpected EOFcan't rewind output file to rewrite SND header%.62s - File created by Sound ExchangeNIST_1ASphere header does not begin with magic mord 'NIST_1A'end_headsample_codingsample_rate sample_byte_formatError reading Sphere header%s %s %s01%s %s %ldSPHERE format only supports ulaw and PCM data.File must be seekable for sphere file outputNIST_1A 1024 sample_count -i %ld sample_n_bytes -i %d channel_count -i %d sample_byte_format -s2 %s sample_rate -i %ld sample_coding -s3 pcm end_head sample_coding -s4 ulaw Could not rewird output file to rewrite sphere header. RRFound header filetype %sLM8953Invalid sample rate identifier found %dSample rate = %ldInvalid filetype ID in input file header, != LM8953txw input file must be a file, not a pipetx16w selected outputtx16w is overriding output format to 1 channel.tx16w is overriding output format to size Signed Word format.Output .txw file must be a file, not a pipetx16w:output finishedSound too large for TX16W. Truncating, Loop Off Creative Voice FileUnsupported VOC format CRLADPCM3 %dUnknown VOC format %dUnsupported VOC format CRLADPCM4 %dUnsupported VOC format CRLADPCM4A %dUnsupported VOC format CRLADPCM2 %dInput .voc file had no sound!VOC file header incorrectunexpected EOF in VOC headerVOC input: short fileOutput .voc file must be a file, not a pipeFile %s: only interpret 8-bit data!File %s: sample rate codes differ: %d != %dFile %s: Sample rate is zero?File %s: Silence sample rate is zeroFile %s: skipping repeat loopDon't understand size %dFile %s: skipping unknown block code %dA(A)A)A*RA*fA*A*A*A+@A/A/@Input not an Ogg Vorbis audio streamWarning: hole in stream; probably harmless Channels: %d Rate: %ld Error writing headre for Ogg Vorbis audio streamError setting up Ogg Vorbis encorder - make sure you've specied a sane rate and number of channelsbase iCoefs differ in %d/14 positionswave header error: wExtSize(%d) too small for nCoefs(%d)ADPCM file nCoefs (%.4hx) makes no sense format[%s]: samplesPerBlock(%d) incompatible with blockAlign(%d)Can only handle 4-bit MS ADPCM in wav filesformat[%s]: expects wExtSize >= %dformat[%s]: expects samplesPerBlock(%d) = %dformat[%s]: expects blockAlign(%d) = %dCan only handle 4-bit IMA ADPCM in wav fileswave header error: wExtSize inconsistent with wFmtLenwave header missing FmtExt chunkUser options overriding encoding read in .wav headerSorry, this WAV file is in IEEE Float format.Sorry, this WAV file is in OKI ADPCM format.Sorry, this WAV file is in Digistd format.Sorry, this WAV file is in Digifix format.Sorry, this WAV file is in Dolby AC2 format.Sorry, this WAV file is in Rockwell ADPCM format.Sorry, this WAV file is in Rockwell DIGITALK format.Sorry, this WAV file is in G.721 ADPCM format.Sorry, this WAV file is in G.728 CELP format.Sorry, this WAV file is in MPEG format.Sorry, this WAV file is in MPEG Layer 3 format.WAV file has unknown format type of %xSorry, this WAV file is in G.726 ADPCM format.Sorry, this WAV file is in G.722 ADPCM format.WAVE file fmt chunk is too shortPremature EOF on .wav input fileerror allocating memory for gsm buffererror setting gsm_option for WAV49 format. Recompile gsm library with -DWAV49 option and relink soxcannot create GSM objecterror during gsm decodeinvalid wav gsm frame size: %d bytesChunk %sWAVE file has missing %s chunkWAVE: RIFF header not foundWAVE header not foundfmt WAVE file is in unsupported Microsoft Official Unknown format.User options overriding channels read in .wav headerUser options overriding rate read in .wav headerSorry, don't understand .wav sizeUser options overriding size read in .wav headerReading Wave file: %s format, %d channel%s, %d samp/sec %d byte/sec, %d block align, %d bits/samp, %u data bytesLISTINFOICRDISFTcue noteadtlltxtlabl %d Extsize, %d Samps/block, %d bytes/block %d Num CoefsGSM .wav: %d Extsize, %d Samps/block, %d samples %d Extsize, %d Samps/block, %d bytes/blockLength in output .wav header will be wrong since can't seek to fix itOverriding output size to bytes for compressed data.Do not support %s in WAV files. Forcing to Signed Words.factWriting Wave file: %s format, %d channel%s, %d samp/sec %d byte/sec, %d block align, %d bits/sampFinished writing Wave file, %u data bytes %u samples GSM6.10 format: %u blocks %u padded samples %u padded data bytes help ! internal inconsistency - data_written %u gsmbytecount %uChannels(%d) must be <= 16 Overriding GSM audio from %d channel to 1 Do not support %s with 8-bit data. Forcing to unsignedDo not support %s with 16-bit data. Forcing to Signed.Can't rewind output file to rewrite .wav header.Microsoft Official UnknownMicrosoft PCMMicrosoft ADPCMIEEE FloatMicrosoft A-lawMicrosoft U-lawOKI ADPCM format.IMA ADPCMDigistd format.Digifix format.Dolby AC2GSM 6.10Rockwell ADPCMRockwell DIGITALKG.721 ADPCMG.728 CELPMPEGMPEG Layer 3UnknownG.726 ADPCMG.722 ADPCMOnly PCM SupportedAC`ACjALALAMAMALAM AMAMAMAMAMAMAMAMAMAM&AMAMAMAM6AM@AMAMAMAMAMAMAMAMAMAMAMAMAMAMAMAMAMAMAMAMAMAMAMAMAMAMJAMTAMAMAMAMAMAMAMAMAMAMdAMnAMAMAMAMxAMAMAMAMAMAMAMAMAMAMAMAMAMAMAMAMAMAMAMAMAMAMAMAMAMAMAMAMAMAMAMAMAMAMAMAMAMASASAWAWASAWAXAXfA\*A\4A\<A\DA\A\A\LA\TA\A\A\A\A\A\A\A\A\\A\dA\A\A\A\lA\tA\A\A\A\A\A\A\A\A\A\A\A\A\A\A\A\A\A\A\A\A\A\A\A\A\A\|A\A\A\A\A\A\A\A\A\A\A\A\A\A\A\A\A\A\A\A\A\A\A\A\A\A\A\A\A\A\A\A\A\A\A\A\A\A\A\A\A\A\A\A\A\A\A\A\A\A\A\A\A\ALawSoundFile**Found Psion magic wordFound PSION magic wordWVE must only supports 1 channel. OverridingWVE must use 8000 sample rate. OverridingWrong version in Psion headerFound inverted PSION magic word. Swapping bytes.Psion header doesn't start with magic word Try the '.al' file type with '-t al -r 8000 filename'avg can only take up to 16 pan valuesUsage: avg [ -l | -r | -f | -b | n,n,n...,n ]Can't average %d channels into %d channelsYou must specify at least one mix level when using avg with an unusual number of channels.Output must have different number of channels to use avg effectInput must have at least two channels to use avg -lInput must have at least two channels to use avg -rInput must have at least four channels to use avg -fInput must have at least four channels to use avg -bOutput must have known number of channels to use avg effectAdAfPAfxAfAfUsage: band [ -n ] center [ width ]Band: center must be < minimum data rate/2 bandpass: illegal bandwidthbandpass: illegal frequencyUsage: bandpass FREQUENCY BANDWIDTHbandreject: illegal bandwidthbandreject: illegal frequencyUsage: bandreject FREQUENCY BANDWIDTHUsage: chorus gain-in gain-out delay decay speed depth [ -s | -t ]chorus: to many delays, use less than %i delaysUsage: chorus gain-in gain-out delay decay speed [ -s | -t ]chorus: warning >>> gain-out can cause saturation or clipping of output <<<chorus: decay must be less that 1.0! chorus: decay must be positive! chorus: delay must be less than 10.0 msec! chorus: delay must be more positive! chorus: speed must be less than 5.0 Hz! chorus: speed must be more than 0.1 Hz! chorus: delay must be less than 100.0 msec! chorus: delay must be more than 20.0 msec! chorus: gain-out must be positive! chorus: gain-in must be less than 1.0! chorus: gain-in must be positive! chorus: Cannot malloc %d bytes! Out of memory,-infInput signals of zero level must always generate zero outputcompander: Odd number of transfer function parameters Each input value in dB must have a corresponding output valuecompander: Odd number of attack & decay rate parametersWrong number of arguments for the compander effect Use: {,}+ {,}+ [ [ [>> gain-out can cause saturation of output <<<echo: decay must be less than 1.0! echo: decay must be positive! echo: delay must be less than %g seconds! echo: delay must be positive! echo: gain-in must be positive! echo: gain-in must be less than 1.0! echo: Cannot malloc %d bytes! Usage: echos gain-in gain-out delay decay [ delay decay ... ]echos: to many delays, use less than %i delaysechos: warning >>> gain-out can cause saturation of output <<<echos: decay must be less than 1.0! echos: decay must be positive! echos: delay must be less than %g seconds! echos: delay must be positive! echos: gain-in must be positive! echos: gain-in must be less than 1.0! echos: Cannot malloc %d bytes! %1[qhltp]Usage: fade [ type ] fade-in-length [ stop-time [ fade-out-length ] ] Times in seconds. Fade type one of q, h, t, l or p. Fade: End of fade-in should not happen before beginning of fade-outFade: warning: End time passed end-of-file. Padding with silenceAAAAAAAAAAAAAfilter opts: %d-%d, window-len %d, beta %f Usage: filter low-high [ windowlength [ beta ] ]Usage: filter low-high [ windowlength ]filter: window length (%ld) <4 is too shortfilter: low(%d),high(%d) parameters must satisfy 0 <= low <= high <= %dfilter: adjusted freq %d-%d is identityfilter: Unable to make high filter filter: Unable to make low filter drain overran obuf by %d Usage: flanger gain-in gain-out delay decay speed [ -s | -t ]flanger: Cannot malloc %d bytes! flanger: warning >>> gain-out can cause saturation or clipping of output <<<flanger: decay must be less that 1.0! flanger: decay must be positive! flanger: speed must be less than 2.0 Hz! flanger: speed must be more than 0.1 Hz! flanger: delay must be less than 5.0 msec! flanger: delay must be positive! flanger: gain-out must be positive! flanger: gain-in must be less than 1.0! flanger: gain-in must be positive! Usage: highp cutoffHighpass: cutoff must be < sample rate / 2 (Nyquest rate) highpass: illegal frequencyUsage: highpass FREQUENCYUsage: lowp cutoffLowpass: cutoff must be < sample rate / 2 (Nyquest rate) lowpass: illegal frequencyUsage: lowpass FREQUENCYMap effect takes no options.Loop info: Loop %d: start: %6d length: %6dMIDI note: %d MIDI low : %d MIDI hi : %d Mask effect takes no options.Usage: pan direction (in [-1.0 .. 1.0])PAN onto a mono channel...PAN cannot handle different rates (in=%ld, out=%ld) use resample or rateunexpected number of channels (in=%d, out=%d)PAN clipped %d values, maybe adjust volume?Usage: phaser gain-in gain-out delay decay speed [ -s | -t ]phaser: Cannot malloc %d bytes! phaser: warning >>> gain-out can cause saturation or clipping of output <<<phaser: warning >>> gain-in can cause saturation or clipping of output <<<phaser: decay must be less that 1.0! phaser: decay must be positive! phaser: speed must be less than 2.0 Hz! phaser: speed must be more than 0.1 Hz! phaser: delay must be less than 5.0 msec! phaser: delay must be positive! Usage: pitch shift width interpole fade (in cents, in ms, cub/lin, cos/ham/lin/trap) (defaults: 0 20 c c)malloc failed in st_pitch_startPITCH cannot handle different channels (in=%ld, out=%ld) use avg or panPITCH cannot handle different rates (in=%ld, out=%ld) use resample or rateunexpected PITCH_FADE parameter %dPITCH clipped %d values... adjust volume with -v option maybe?AA0A0A0A0AA0A0A0AxA0A0A0A0A0A0A0AA0A0A0A0A0A0A0A0A0A0A0A0A0A0AA0A0A0A0AA0A0A0AxA0A0A0A0A0A0A0A-width-cutofflongPolyphase: unknown argument (%s %s)!factors(%d) =Number %d too large of a prime. Illegal buffer %p or length %d to nuttall. Illegal buffer %p or length %d to hamming. Illegal buffer %p, length %d, or cutoff %f. Poly: input rate %d, output rate %d. %d stages.nutPoly: window: %s size: %d cutoff: %f.Poly: stage %d: Up by %d, down by %d, i_samps %d, hsize %dPoly: : filt_len %d, cutoff freq %.1fPoly: output samples %d, oskip %dhamInput and Output rate must not be the same to use polyphase effect %)+/5;=CGIOSYaegkmq %379=KQ[]agou{  #-39;AKQWY_eikw)+57;=GUY[_mqswRate effect takes no options.rate effect can only handle rates <= 65535rate effect reduces data to 16 bitsInput and Output rates must be different to use rate effectresample: rolloff factor (%f) no good, should be 0.01>> gain-out can cause saturation of output <<<reverb: gain-out must be positivereverb: delay must be positive! reverb: Cannot malloc %d bytes! Reverse effect takes no options.Reverse effect can't create temporary file Reverse effect write error on temporary file Internal error: reverse_flow called in wrong phaseReverse effect read error from temporary file Reverse effect finds odd temporary file %lf%csilence threshold should be less than 0.0 dBsilence threshold should be between 0.0 and 100.0 %%Invalid unit specifiedUsage: silence above_periods [ duration thershold[d | %% ] ] [ below_periods duration threshold[ d | %% ]]Periods must not be greater then zeroPeriods must not be negativeUnable to allocate memoryB BPBBXB`n = %d cent = %d speed = %f speed [-c] factor (default 1.0, <1 slows, -c: factor in cent)malloc failedSPEED: %d values clipped...-s option: invalid argument-rms-freqSummary effect: unknown optionFFT can only use sample buffers of 2^n. Buffer size used is %ld Unable to allocate memory for FFT buffers. %08lx %f %f Samples read: %12u Length (seconds): %12.6f Scaled by rms: %12.6f Maximum amplitude: %12.6f Minimum amplitude: %12.6f Midline amplitude: %12.6f Mean norm: %12.6f Mean amplitude: %12.6f RMS amplitude: %12.6f Maximum delta: %12.6f Minimum delta: %12.6f Mean delta: %12.6f RMS delta: %12.6f Rough frequency: %12d Try: -t raw -b -u Try: -t raw -b -s Try: -t raw -b -U Can't guess the type Probably text, not sound Volume adjustment: %12.3f Scaled by: %12.1f %.3f Usage: stretch factor [window fade shift fading] (expansion, frame in ms, lin/..., unit<1.0, unit<0.5) (defaults: 1.0 20 lin ...) error with shift ratio valueUsage: stretch factor [window fade shift fading] (expansion, frame in ms, lin/..., unit<1.0, unit<0.5) (defaults: 1.0 20 lin ...) error with fading ratio valueUsage: stretch factor [window fade shift fading] (expansion, frame in ms, lin/..., unit<1.0, unit<0.5) (defaults: 1.0 20 lin ...) error while parsing fading ratioUsage: stretch factor [window fade shift fading] (expansion, frame in ms, lin/..., unit<1.0, unit<0.5) (defaults: 1.0 20 lin ...) error while parsing shift ratioUsage: stretch factor [window fade shift fading] (expansion, frame in ms, lin/..., unit<1.0, unit<0.5) (defaults: 1.0 20 lin ...) error while parsing fade typeUsage: stretch factor [window fade shift fading] (expansion, frame in ms, lin/..., unit<1.0, unit<0.5) (defaults: 1.0 20 lin ...) error while parsing window sizeUsage: stretch factor [window fade shift fading] (expansion, frame in ms, lin/..., unit<1.0, unit<0.5) (defaults: 1.0 20 lin ...) error while parsing factorsome malloc failedSTRETCH cannot handle different rates (in=%ld, out=%ld) use resample or rateSTRETCH cannot handle different channels (in=%d, out=%d) use avg or panSTRETCH clipped %d values...Usage: swap [1 2 | 1 2 3 4]invalid swap channel options usedCan't swap channels on mono data.sinecreatesynth: parameter errorUsage:synth [length] type mix [freq[-freq2]] [off] [ph] [p1] [p2] [p3] length in sec or hh:mm:ss.frac, 0=inputlength, default=0 is sine, square, triangle, sawtooth, trapetz, exp, whitenoise, pinknoise, brownnoise, default=sine is create, mix, amod, default=create frequency at beginning in Hz, not used for noise.. frequency at end in Hz, not used for noise.. can be given as %%n, where 'n' is the number of half notes in respect to A (440Hz) Bias (DC-offset) of signal in percent, default=0 phase shift 0..100 shift phase 0..2*Pi, not used for noise.. square: Ton/Toff, triangle+trapetz: rising slope time (0..100) trapetz: ON time (0..100) trapetz: falling slope position (0..100)synth: too many parameterssynth: illegal freq2synth: illegal freqmixamodfmodsquaresawtoothtriangleexptrapetzwhitenoisepinknoisebrownnoisevoicenoisesynth: no type givenCould not allocate memeorysynth: internel error 2synth: internal error 1synth: can not operate with more than %d channelsB>B>B>B>B?B>B?BA2BB$BBJBBZBBBBBC BC*BCBCTrim usage: trim start [length]Usage: vibro speed [ depth ]Vibro: speed must be < 30.0, 0.0 < depth < 1.0Vibro: Cannot malloc %d bytesUsage: vol gain [ type [ limitergain ] ] (default type=amplitude: 1.0 is constant, <0.0 change phase; type=power 1.0 is constant; type=dB: 0.0 is constant, +6 doubles ampl.) The peak limiter has a gain much less than 1.0 (ie 0.05 or 0.02) which is only used on peaks to prevent clipping. (default is no limiter)VOL cannot handle different channels (in=%d, out=%d) use avg or panVOL cannot handle different rates (in=%ld, out=%ld) use resample or rateVOL clipped %d values, amplitude gain=%f too high...VOL limited %d values (%d percent).BLtBLtBLtBLXBLtBLtBLtBLtBLtBLtBLtBLtBLtBLtBLtBMBLtBLtBLtBLtBLtBLtBLtBLtBLtBLtBLtBLtBLtBLtBLtBLtBLtBLtBLtBLXBLtBLtBLtBLtBLtBLtBLtBLtBLtBLtBLtBM3ff3@0MSADPCM bpred >= nCoef, arbitrarily using 0  "%)-27<BIPXakv3Qs Vl$V LLT!%(,1[6K;AHDO~Wq`/itbIMA_ADPCM block ch%d initial-state (%d) out of range $Header: /cvsroot/sox/sox/gsm/gsm_create.c,v 1.1 2001/11/14 03:31:26 cbagwell Exp $BBBBBBNc <= 120 && Nc >= 40long_term.cCalculation_of_the_LTP_parametersscal <= 100 && scal >= -100scal >= 0dmax > 0bcGsm_Long_Term_PredictorNcdppdpbrp != (-32767 - 1)Gsm_Long_Term_Synthesis_FilteringNr >= 40 && Nr <= 120BBnBB>scalauto <= 4lpc.cAutocorrelationsmax > 0*r != (-32767 - 1)Reflection_coefficients*r >= 0temp >= 0 && temp < 32Transformation_to_Log_Area_Ratiostemp >= 11059temp >= 26112BBBBs1 != (-32767 - 1)preprocess.cGsm_PreprocessSO <= 0x3FFCSO >= -0x4000mant >= 0 && mant <= 7rpe.cAPCM_quantization_xmaxc_to_exp_mantexp >= -4 && exp <= 6temp1 >= 0 && temp1 < 16APCM_quantizationexp <= 4096 && exp >= -4096temp <= 11 && temp >= 0exp <= 6 && exp >= 0exp <= 5temp <= 7 && temp >= -7APCM_inverse_quantization*xMc <= 7 && *xMc >= 00 <= Mc && Mc <= 3RPE_grid_positioningB`BTBHB= 0 && denum >= numgsm_div;bٌ8ߐ<APA$B`DBpC0CD$E\IJ0,KPNtN@NO`OOP0P4PPtPRLRSPS T(UPWxX X4XdXY[D\,^\P_t_``abpcHdtf@gi|k$lPl`mnhnoo\opt,u4XvHw0xdxxz<{d{|$|h~H8Dd|\ Hlp8`X$|t$P@\x08HH<Xl|@$$L4lD\tT d48\X 8$@\Dx`(t88Xtx(d8XTxD8X4xH0@X@H t8`xôŌƀư @@ hhʜHˀ8\@h$x(80@8Pp٤(8 `<$(Ppt8tP$ADB,pC<ɘCxɸD|FH0I\I|KdʨLXMN O@Pd`Q˄TlˬUVV(WHBDBHBNBRBVBZB^BdBhBlBpBxB|BӀCp @p@l@@@@@plCqA@@\A@,@x@Q\Cq @@@\@@@@Cq@@QT@QL@@QT@QL@Q\Cq @@@QL@@@@Q\Cq( @@@@@h@@@lCq0@@@x@@@@Q\Cq<@@@@QL@x@(@QL@Q\CqD@P@@x@@@@Q\CqP@@t@p@H@@@Q\CqX@ @@@p@4@װ@Q\Cq`At@@\A@,@x@Q\CqhA@@\A8@,@x@Q\Cqp@D@(@P@x@\@ތ@Q\Cqx@@@,@@@@Q\Cq@@@@@@@d@Q\Cq@fl@@\@f@,@x@Q\Cq@@@\@@@$@lCq @@@\@l@,@x@lCq@@@\@@,@x@Q\Cq A@@\A@,@xAxCq@@@\@@,@x@Q\Cq A Ax@QLAA,AA xCq Al@@\ADA8AhA0CqAA @\ATAA8@Q\Cq@f@kX@l@ld@m @m@Q\Cq@@@\@X@,@x@Q\CqAAAAAA|@Q\Cq@,@@\@h@,@x@Q\CqA @@\AH@,@x@Q\Cr@@@\@@,@x@Q\CrA!A#A%A%A&A'\@Q\CrA/A2A3dA4A6(A7@Q\Cr A@AM@QLAPHAYAZA\Cr$A]A`@\A_pA`<A`xA]xB*Aa@AcAf@QlAhxBӄAhAiPAk@QlAkBӊAkAlHAo4@Ql@QdBӔAmhAnAo4@Ql@QdBӠAp(AqAu`AvAxBӨAxlA|A~AxABӰAxAA@Ql@QdBӶA AA@QlA,BӾAAA@QlAlBAtAA<AABAAAAABA0A4APAA8BAtAAAAhBAAA$AAPBAhA`A4AA@BAAA@QlABAA Ao4@Ql@QdBALAAd@QlABA AAo4@Ql@QdBAAAh@Ql@QdBAp@QdA@Ql@QdCAA\A @QlALBAŒAÄAhAAxB A͌ADAAPAHB&A֬AAAACA0AhA@Ql@QdB.AAAAdA$B0AdAHBBBB8BB,BBBB@BLB B BpBBHBPBLB`BTB0BN BBBBB!TBTB)B+|B.B1B1B\B2HB3B4B5B5BbB8B>BDBEBEBhBEBFBG@Ql@QdBnBHBJpBK4@Ql@QdBtBKBM|BNL@QlBO|BxCCbBԂBԊBԔBԊBԊBԊBԚBxBԦB԰BBBBBBBBB:p:9Dn74; m;f<Y<#6; <^P<=@=Nϧ=B=~=w>+!>I>H)>cm>#?d?Y|?tߋ?Y|?d>#>cm>H)>I>+!=w=~=B=Nϧ=@<<^P; <#6<Y;f; mҹ749Dn::p:$;b;tG<]< m<<.,.Ĥ>P>Ⱥb? 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3.3GCC: (GNU) 3.3GCC: (GNU) 3.3GCC: (GNU) 3.3GCC: (GNU) 3.3GCC: (GNU) 3.3GCC: (GNU) 3.3GCC: (GNU) 3.3GCC: (GNU) 3.3GCC: (GNU) 3.3GCC: (GNU) 3.3GCC: (GNU) 3.3GCC: (GNU) 3.3GCC: (GNU) 3.3GCC: (GNU) 3.3GCC: (GNU) 3.3GCC: (GNU) 3.3GCC: (GNU) 3.3GCC: (GNU) 3.3GCC: (GNU) 3.3GCC: (GNU) 3.3GCC: (GNU) 3.3GCC: (GNU) 3.3GCC: (GNU) 3.3GCC: (GNU) 3.3GCC: (GNU) 3.3GCC: (GNU) 3.3GCC: (GNU) 3.3GCC: (GNU) 3.3GCC: (GNU) 3.3GCC: (GNU) 3.3GCC: (GNU) 3.3GCC: (GNU) 3.3.shstrtab.interp.note.ABI-tag.hash.dynsym.dynstr.gnu.version.gnu.version_r.rela.dyn.rela.plt.init.text.fini.rodata.eh_frame_hdr.data.eh_frame.dynamic.ctors.dtors.jcr.got.sbss.bss.comment @ @$$ !@DDx' @/@ l l7o@Do@`S@TTT ]@  g@\b@ `m@&\&\PsBˬˬ8yBCMMCp`uCxxPD6&D7'D7'D7'D7'D9T)XD9X)XL)X/Xusr/doc/0040755000000000000000000000000010032107730011127 5ustar rootrootusr/doc/sox-12.17.4/0040755000000000000000000000000010032107730012550 5ustar rootrootusr/doc/sox-12.17.4/Changelog0100644000000000000000000006750707605177164014422 0ustar rootrootChange History -------------- This file contains a list of all changes starting after the release of sox-11gamma. sox-12.17.4 ----------- o Peter Nyhlen fixed a problem with reading Comments in Ogg Vorbis files. o Added install target to allow installing libgsm from main Makefile. Leigh Smith. o Minor updates to sox.c to free unused memory and close all input files during failures. o Pieter Krul added a patch that makes play script look at AUDIODEV environment variable if it exists to find which device to use. This allows scripts to work with Solaris SunRays and is a good idea in general. o Updated config.sub to detect latest supported OS's. o Fabrizio Gennari added support for reading and writing MP3 files using the external libraries libmad and libmp3lame. o Jens Henrik Goebbert sent in several bugfixes for integer overflows in the compand effect. o Dan Dickerman sent in patches for integer overflows in the resample effect. o Jimen Ching sent in a fix for multi-channel sound file processing using the avg effect. o Richards Bannister added patches to clean up prototypes and filter private sizes being to small. o Jimen Ching adds -d option to specify 64bit data size and changed Ulaw/Alaw encoding to default to 8bit data size if not specified. o David Singer pointed out that a MS program creates AIFF files with an invalid length of 0 in its header. Changed SoX to warn the user but continue instead of aborting since SoX can still read the file just fine. o Bert van Leeuwen added a file handler for Psion record.app used for System/Alarms in some Psion devices. o Richard Bannister sent in a patch to make writing vorbis files work with Vorbis 1.0 libraries. o Fixed configure scripts so that they can be ran with the --with-oss-dsp, --with-alsa, and --with-sun-audio options. Was causing compile time problems. Reported by Raul Coronado. o Change Ogg Vorbis support to use VBR encoding to match defaults of oggenc based on suggestion from Christian Weisgerber. o Prints error message now when a channel value of -1 is given. Reported by Pierre Fortin. o Fixed bug were memory could be trashed if a input WAV file contained a comment. Found by Rhys Chard. o Change command line to compile soxmix.o slightly to try and make Forte compiler happy. o Added support for ALSA 0.9 driver. Jimen Ching sox-12.17.3 ----------- o Removed check that prevented pan from being invoked when the input and output channels were the same. o Ciaran Anscomb added a flush to sunaudio driver after changing settings. This is because it can start buffering data as soon as the device is open and the buffered data can be in a wrong format. o trim wasn't accounting for # of channels and was generally broken. o Jeff Bonggren fixed trim bugs were it was failing when triming data that equaled to BUFSIZ. Also, trim now immediately returns ST_EOF when its done instead of requiring an extra call that returns no data. o auto effect wasn't rewinding the file if the file was less then 132 bytes. Changed auto parsing of header to be incremental instead of reading in a large buffer. o William Plant pointed out a bad pointer access in fade effect's parsing of options. o Ken pointed out a problem were private data was not 8-byte aligned and causing crashes on most RISC CPU's. Fixed by going back to old style of declaring private data as type "double" which usually forces strictest alignment. o ima_rw was miscompiling on alpha's because of a header ordering problem. o Erik de Castro Lopo pointed out that when writing 16-bit VOC files the headers did not contain the correct length or encoding type. o Seperated st.h into 2 files. st.h for libst users and st_i.h for internal use. o Added new types used extensively by libst: st_sample_t & st_size_t. This allows for more deterministic behavior on 64-bit machines and also allows sox to possibly work with much larger file sizes. o SoX was some times getting confused and thinking an EOF was an error case when reading audio files. Removed unneeded aborts when EOF was OK. o Silence effect was broken on stereo files. Also, made thresholds relative to original bit percision of audio data. When 16-bit audio is scaled up to 32-bits, a little bit of noise starts to look like a large amplitude of noise. Also, now using RMS values to smooth out clicks. RMS rolling window size is 1/10 of sample rate. o Changed Floats into a type of encoding instead of a size of audio data. o Put a flush at the end of OSS driver so that no old data would be left in internal buffers after changing audio format parameters. o Fixed problem were play script wasn't installed correctly if you build from another directory (pointed out by Mike Castle). o Made GSM support internal to libst (no external library required). o Change configure script to enable ulaw/alaw lookup tables and GSM support by default. Also have Makefile's make use of more configure prefix options to allow for customized installs. o Reverted ulaw/alaw conversion routines back to Sun's versions. o Modified raw file handler to write files in the same generic buffered fashion that was added for reading in 12.17.2. Seems to have speed up some types of writing. o Reading Ogg Vorbis files could get confused of when EOF was reached. o Added uninstall rules to Makefile. Added new ststdint.h to define *int*_t typedefs. o Added internal strcasecmp for OS/2. o Added support for swapping "bit" order (MSB becomes LSB) for raw u-law and A-law data. Some ISDN equipment prefers it this way. Use -x flag or new .la or .lu file extensions. o Annonymous patch submitted to fix types and spelling problems in various files. Also, updated VOC files to have u-law and A-law support as well as able to read in VOC files using a pipe. More examples added to soxexam file. sox-12.17.2 ----------- o Daniel Culbert found and fixed a bug in the polyphase effect that occurs on platforms that rand() can return large values. The bug resulted in polyphase resampling an audio file to a different rate then it said it was. o Stan Seibert contributed a handler for Ogg Vorbis files. It handles all input formats but can only save using default settings. o Darrick Servis has made major cleanups in the code in regards to error conditions. Helps people using libst. o Darrick Servis has added added optional seek functionality sox. Several formats have been modified to make use of this. o Geoff Kuenning rewrote the average effect into a general-purpose parametric mapping from N channels to M channels. o Geoff Kuenning added an optional delay-time parameter to the compander effect to allow companding to effectively operate based on future knowledge. o Geoff Kuenning Added support to fade and trim effect for specifying time in hh:mm:ss.frac format. Fixed a bug that caused integer overflow when large start/stop times were used. o Geoff Kuenning updated play/rec/soxeffect scripts to handle all effects added since 12.17. Spell-checked soxexam.1 file. o Jimen Ching updated ALSA configure support to auto-detect 4.x or 5.x API and compile correctly under those two. All other versions are unsupported. o Merged in the NetBSD package changes into CVS finally. o Removed broken support for non-ANSI compilers. o Makefile now places the correct path to SoX in the play/rec scripts based on configuration script values. o Alexander Pevzner provided a fix for OSS driver for sound being dropped under heavy CPU loads. Moved GETBLKSIZE operation until after setting up the format (SBLive! was modify the block size after changing formats). o With help from David Blythe, updated OSS drivers to use newer format interface. OSS driver will now attempt to detect a valid endian type to use with sound card. o Carsten Borchardt pointed out a bug in lowp filter. Added new nul file handler that reads and writes from/to nothing. Also added new synth effect that creates sounds using a simple synthesizer. Created a testcd.sh that uses two new features to create a test sound CD for testing audio equipment. o Ben Last added a new program that uses libst and will merge two seperate audio files into a single file with multiple channels. This was merged into the standard sox.c file by cbagwell. o Andreas Menke fixed some problems with the speed effect and how effects were drained. Also improved the usage of printf()'s to use stderr. o Corrected AU header length value when comments were less than 4 bytes. o Added support for reading non-standard bit size data from AIFF files. o Ignore unmatched MARK/INSTR chunks in AIFF files now instead of quiting. o Fixed ALAW encoding bug in .au files as pointed out by Bruce Forsberg. o Unified the raw reading functions. Probably slightly faster for most datatypes but was done to fix recording from the OSS driver. Control-C stopped working somewhere during the 12.17 series. o Ford Prefect added a dcshift which can shift the midline amplitude towards the true center. This will allow for a greater range of volume adjustments without clipping audio data. o Heikki Leinonen submitted a silence effect that will trim off silence from the beginning of a file. cbagwell made some modifications to trim off front and back as well as some other tweaks. o Made the "auto" effect the default file handler for input files. Also have auto handler now use file extensions if it can't figure it out. sox-12.17.1 ----------- o Andreas Kies fixed a bug were we were not detecting correctly if an output file was seekable. o Fixed a bug in the mask effect introduced in 12.17. If the libc version of rand() returned more then 15-bit values then it would trash your data. Reported by Friedhel Mehnert. o Added a new fade in/out effect from Ari Moisio. o AIFF files now ignore a MARK chunk if the loop type is NoLoop (0). o Fixed bug were it was impossible to output ADPCM data in wav files. o Fixed bug were rate had to be specified for sphere files (fix from Antti Honkela). o Added small work around to let compile with cygwin's gcc 95.2 which also now allows to compile with GSM support under windows. o Removed accessing of sound devices in tests for sound support and instead just look for needed header files. This allows the sound support to be detected even if the device is currently busy or when compiled on a box that doesn't have a sound card but the OS supports it (which is the enviornment of most distributions creating new binaries). o Added support to partially handle AIFC files but only uncompressed versions. This should allow people to work with raw CD audio data on Mac OSX and also gives a basis for adding future support for things like ADPCM processing. o Added new "earwax" effect from Edward Beingessner. It is meant to be used for CD audio played through headphones. It will move the sound stage from left/right to in front of you. o Trim effect did not compute locations as was documented in the man pages. Changed effect so that it computed the time the same way that the fade effect does. sox-12.17 --------- o Sox can now read and write w98 compatible gsm .wav files, if compiled properly with libgsm. Thanks go to Stuart Daines for the gsm-wav patches. This is new, and relatively untested. See -g format option. o Sox can now write IMA_ADPCM and ADPCM compressed .wav, this is new, and relatively untested. See -i and -a format options in manpage. o General changes to wav.c for writing additional wav formats. Reading wave headers: more consistency checks. Writing wave headers: fixes for w98. o Speedups to adpcm read routines, new codex versions are now in ima_rw.c and adpcm.c. o Speedups for raw.c, especially for gcc with glibc. o Fixed a segfault problem with ulaw/alaw conversion, where an out-of-range index into the tables could occur. o More detailed output from the stat effect. o Continued rewrite of resample.c, now it is almost as fast with floating arithmetic as the old (buggy) version was with 16-bit integer arithmetic. The older version in sox-12.16 shifted frequencies slightly and was less accurate. o Extensive rewrite of polyphas.c, should be faster and use less memory now. The sox-12.16 polyphase code had some bugs. o New effect 'filter' which is a high-quality DSP lowpass/ highpass/bandpass filter using windowed sinc function methods, like polyphase and resample. o Jan Paul Schmidt added new low/high/bandpass and bandlimit filters to sox. They have much better results then the old versions of low/high/bandpass. The new effects are all Butterworth filters. o New data file type supported, -sl or extension .sl for signed 32-bit integers. Some simplification of the raw.c source. o Some test programs and scripts in the test directory, for making gnuplot plots of frequency response, error-levels of rate-conversion and filter effects. o Removed sbdsp code. All modern unixes are support via OSS, ALSA, or sun audio device interfaces. o Added AVR handler from Jan Paul Schmidt. o SoX now waits until the last possible moment before opening the output file. This will allow all input and effect options to be parsed for errors and abort before overwriting any file. o SoX will no longer write to files that exists. This will keep it from deleting files when a user mistakenly types "sox *.wav". o Added new compander effect from Nick Bailey. Nice general purpose filter. o Under Solaris, SoX now checks hardware ability to play stereo/PCM and forces output data to match. Sorry, no SunOS support. I don't have access to one any more. o Fixed array overrun bug in rate effect as pointed out by Ian Donaldson. o Fixed clip24() range as pointed out by Ted Powell. o Fixed possible segfault in echos effect, as pointed out by Zinx Verituse. o Moved most documentation to new "soxexam.1" manual page so that all users on a unix system will have access to important information. This means no more TIPS, CHEATS, or CHEATS.eft files. o Richard Kent fixed a byte alignment problem in aiff comment code. o Loring Holden added support to create missing directories on install and support for installs from outside the source directory. o Fabien COELHO added a pan and volume effect. o Fabien COELHO added a speed effect to sox (like speeding up a tape machine). Also added pitch which changes pitch without effecting duration and stretch which stretch time without effecting pitch. o Leigh Smith updated aiff support to use the COMT check when writing comments instead of the ANNO. It is the prefered method from Apple and newer programs are starting to require it. Also fixed some bugs in how we compute the length of various chunks in output files. o Changed the default block alignement for IMA ADPCM WAV files to use 256 which is what windows programs use. Badly written readers expect 256. o Matthias Nutt helped add support for specifying multiple effects to SoX on the command line. o Curt Zirzow added a trim effect to trim off audio data. o Updated ALSA driver to support new interface. Jimen Ching sox-12.16 --------- o Changed raw data functions to perform I/O operations using block reads. Should improve speeds greatly, especially when filesize is in megs. Got code ready to tweak speed more which also forced me to clean up Endian test code. o Fixed a bug in .au's handling of G.723. It wasn't using the correct number of bits. Added A-law support to .au. o Quoted $filename in play/rec scripts so that files with spaces in their names can be given. o Old OS/2 support didn't work. Replaced with known working EMX GCC compatible code. o ADPCM WAV files were defaulting to 8-bit outputs and thus losing some persision. Now defaults to 16-bit signed uncompressed data. o Fixed a couple cross-platform compiler issues. o Jimen Ching has added support for "configure" in to SOX. Finally, a good solution to cross-platform compiling! o Jimen Ching has added native support for the ALSA driver to play and record audio from. (jching@flex.com) o Minor correction for -r example in manual page. o Renamed sox.sh to soxeffect and rewrote. Symbolic links can be made from this file to the name of a sox effect. It will then run that effect on STDIN and output the results to STDOUT. o Fixed up some makefiles and 16-bit support from patches sent by Mark Morgan Lloyd (markMLl.in@telemetry.co.uk). Also added some nice DOS test bat files from him as well. o Cleaned up some more cross-platform compile problems. In the process got it working with Turbo C again, kinda. It still locks DOS up at times. o Made AIFF handler work with invalid headers that some programs generate. Also fix an Endian bug thats been there for quite a long time (when ran on Intel machines). Made comment lengths even length to make most MacOS programs happy. cbagwell o Resample function was updated by Andreas Wilde (andreas@eakaw2.et.tu-dresden.de) to fix problem were freqs. were off by a factor of 2. o Added an effect that swaps stereo channels. cbagwell o Combined play and rec scripts to a single program to ease mantaince. Also added GNU style long-options (--help). Careful, some options have change (but more logical). o Fixed a very old silly bug were low/high/bandpass filters would add some trash data to the end of the sound file. o "avg" effect wouldn't give error message when you tried to average x number of channels in to the same number of channels. o Fixed core dump when writting AIFF files with no comments. (Olaf Pueschel) sox-12.15 --------- o Juergen Mueller moved Sox forward quite a bit by adding all the most commonly known "Guitar Effects". He enhanced echo support, added chorus, flanger, and reverb effects. He also wrote a very handy CHEAT.eft file for using various effects. o Incorporated Yamaha TX-16W sampler file support provided by Rob Talley (rob@aii.com) and Mark Lakata (lakata@physics.berkeley.edu). o Fixed a small bug in hcom compression, dependent on sign extension. Leigh Smith (leigh@psychokiller.dialix.oz.au). o sox -h now prints out the file formats and effects supported. Leigh Smith and Chris Bagwell. o smp transfers comments more completely. Leigh Smith. o aiff manages markers and loops correctly and produces more verbose output. Leigh Smith. o Added polyphase resampler (kb@ece.cmu.edu). This adds a slightly different resampling algorithm to the mix. o Michael Brown (mjb@pootle.demon.co.uk) sent a patch to stop crashes from happening when reading mono MS ADPCM files. o Fabrice Bellard has added a less buggy 'rate' conversion. I've left the old rate code included but if all goes well this will become the new 'rate'. Please test and let me know how it works. Resample effect needs to be reworked now. o Heiko Eissfeldt: Implemented a simple deemphasis effect for certain audio cd samples. o Matija Nalis (mnalis@public.srce.hr) sent a patch to fix volume adjustment (-v) option of sox. o Fixed typo in optimazation flag in unix makefile, as pointed out by Manoj Kasichainula (manojk@io.com). o Fixed missing ';;' in play script. cbagwell o Fixed bug in determining length of IMA and MS ADPCM WAVE files. cbagwell o Fixed bug in how stereo effects were drained which fixed the "reverse" effect from only saving half of stereo files. cbagwell o Can use "-e" without an effect again. o Added -g and -a options for new style support of GSM and ADPCM. Added error checking to various formats to avoid allowing these types. sox-12.14 --------- o Bumped major version number up and shortened name. The shorter name should help the various distributions using this package. o Added support for MS ADPCM and IMA (or DVI) ADPCM for .wav files. Thanks to Mark Podlipec's xanim for this code (podlipec@ici.net). o Change Lance Norskog's email address to thinman@meer.net. The old one was bouncing. o Added path string to play and rec strings so that it could be run by users without complete paths setup (i.e. Ran by "rc" files during bootup or shutdown) o Fixed -e option from Richard Guenther (richard.guenther@student.uni-tuebingen.de) and fixed a small bug in stat. o Fixed a bug in the mask effect for ULAW/ALAW files. o Fixed a bug in cdr output files that appended trash to end of file. o Guenter Geiger (geiger@iem.mhsg.ac.at) made a rather large patch to allow sox to work on 64-bit alphas. It was done the easiest meathod by changing all long declarations to use a macro that knows to make it 32-bits. Want to port to another 64-bit-but-not-alpha machine? Grep for "alpha" to see changes. There are most likely several bugs left for alphas. Guenter is also supporting this package for the Debian distribution. o Did some major code cleanups to clear out some warning messages during compile. This is to clear up problems I'm finding under both alpha and dos. Some warning messages are actually useful now (pointing out possible data loss). Hopefully, I didn't break anything. o Code clean up allows me to finally compile code under Turbo C again. Too bad the EXE gets a currupted stack somewhere and locks up the system. Anyone want to try it with Borland C for me? If you get a working EXE I would like to start distributing a DOS package like there used to be. o Speaking of cleanups, anyone want to help cleanup the makefiles for various platforms? They are quite outdated right now and it is very obvious that Sox hasn't been able to compile under all the platforms it once did for several releases. Please send in the cleaned-up makefile versions along with what programs you used to compile it with. o There is a known bug in hcom's compress() function. It is allocating memory that can't be free'd under some OS's. It causes a core dump. sox-11gamma-cb3 --------------- This release of sox is mainly a bugfix release. The following things have changed: o Documentation has been updated when it was obviously wrong. Much more work could be done. Man pages were updated to work correctly on Solaris and add some missing info. o Several people sent me patches to fix compiling on Solaris as well as fix a few bugs. o Change USS driver's name to OSS. Man, does that driver like to change names! This could cause problems if you have made your own custom play and rec scripts. o Updated my email address. Sorry if I haven't responded to any emails as I no longer have access to my old address. Please use cbagwell@sprynet.com. o Fixed unix test scripts so that they worked again. o Fixed endian bug in psion .wve code. o Replaced outdated voc info file with detailed format info inside voc code. o Added new sound format, cvsd (Continuously Variable Slope Delta) from Thomas Sailer (sailer@ife.ee.ethz.ch). sox-11gamma-cb2 --------------- This release of sox is based on the latest gamma version released plus some patches I've made to support the following new features: I would like to thank everyone that wrote me about the long standing bug in Sox that could DELETE your /dev/* file if the program was aborted for reason such as invalid audio file. Special thanks for Bryan Franklin for sending in a patch when I was to busy to even look for it. o Better play support for 8-bit stereo voc files. New support for outputing both 8-bit and 16-bit stereo voc files. o Built-in support for playing and recording from Linux /dev/dsp. This is a re-write and seperate module from the previous support included inside the sbdsp module. Also fixes a buffer size bug that showed up when using newer versions of OSS. This driver will work with OSS (and older versions called USS, TASD and Voxware). o Support for audio playing and recording with SunOS /dev/audio. o Fixes a bug were /dev/audio or /dev/dsp could be deleted when playing an invalid format audio file. o Expanded options for play and rec scripts. You can now specify sox effects after the filename and hear them in real time. Please be sure that an older version of sox is not in your path because these script will possibly find it first and incorrectly use it. o Setting play/record volume still requires an external program. If you have one a command line program to do this (such as "mixer" for Linux) then you will want to edit the play and rec to use this. The current support for it is only in example form of how it can be done. The following information originally appeared in the main README file. It has been moved here since all new additions are documented in this file. Creator: Lance Norskog thinman@meer.net (inactive currently) Mantainer: Chris Bagwell cbagwell@sprynet.com Contributors: Juergen Mueller jmueller@uia.ua.ac.be chorus, echo, echos, flanger, phaser, and reverb effects. Guido Van Rossum guido@cwi.nl AU, AIFF, AUTO, HCOM, reverse, many bug fixes Jef Poskanzer jef@well.sf.ca.us original code for u-law and delay line Bill Neisius bill%solaria@hac2arpa.hac.com DOS port, 8SVX, Sounder, Soundtool formats Apollo fixes, stat with auto-picker Rick Richardson rick@digibd.com WAV and SB driver handlers, fixes David Champion dgc3@midway.uchicago.edu Amiga port Pace Willisson pace@blitz.com Fixes for ESIX Leigh Smith leigh@psychokiller.dialix.oz.au SMP and comment movement support. AIFF Loop/MIDI support David Sanderson dws@ssec.wisc.edu AIX3.1 fixes Glenn Lewis glewis@pcocd2.intel.com AIFF chunking fixes Brian Campbell brianc@quantum.qnx.com QNX port and 16-bit fixes Chris Adams gt8741@prism.gatech.edu DOS port fixes John Kohl jtkohl@kolvir.elcr.ca.us BSD386 port, VOC stereo support Ken Kubo ken@hmcvax.claremont.edu VMS port, VOC stereo support Frank Gadegast Microsoft C 7.0 & C Borland 3.0 ports David Elliot CD-R format support David Sears Linux support Tom Littlejohn Raw textual data Boisy G. Pitre boisy@microware.com OS9 port Sun Microsystems, Guido Van Rossum CCITT G.711, G.721, G.723 implementation Graeme Gill graeme@labtam.labtam.oz.au A-LAW format, Good .WAV handling, avg channel expansion Allen Grider grider@hfsi.hfsi.com VOC stereo mode, WAV file handling Michel Fingerhut Michel.Fingerhut@ircam.fr Upgrade 'sf' format to current IRCAM format. Float file support. Chris Knight Achimedes Acorn support Richard Caley R.Caley@ed.ac.uk Psion WVE handler Lutz Vieweg lkv@mania.RoBIN.de MAUD (Amiga) file handler Tim Gardner timg@tpi.com Windows NT port for V7 Jimen Ching jiching@wiliki.eng.hawaii.edu Libst porting bugs Lauren Weinstein lauren@vortex.com DOS porting, scripts, professional use Stan Brooks stabro@megsinet.net Rewrite of resample and polyphase code. DSP filter effect. Some test code/scripts. Stuart Daines Patches for r/w support of gsm-encoded wav files, Cleanup of wav.c. Chris Bagwell cbagwell@sprynet.com OSS and Sun players, bugfixes, ADPCM support, patch collection and maintance. Matthias Nutt Multiple effects from command line. (your name could be here, too) (I've probably lost a few, and several people fixed the same bugs.) usr/doc/sox-12.17.4/Copyright0100644000000000000000000006347407023017252014463 0ustar rootroot GNU LESSER GENERAL PUBLIC LICENSE Version 2.1, February 1999 Copyright (C) 1991, 1999 Free Software Foundation, Inc. 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA Everyone is permitted to copy and distribute verbatim copies of this license document, but changing it is not allowed. [This is the first released version of the Lesser GPL. It also counts as the successor of the GNU Library Public License, version 2, hence the version number 2.1.] Preamble The licenses for most software are designed to take away your freedom to share and change it. By contrast, the GNU General Public Licenses are intended to guarantee your freedom to share and change free software--to make sure the software is free for all its users. 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To apply these terms, attach the following notices to the library. It is safest to attach them to the start of each source file to most effectively convey the exclusion of warranty; and each file should have at least the "copyright" line and a pointer to where the full notice is found. Copyright (C) This library is free software; you can redistribute it and/or modify it under the terms of the GNU Lesser General Public License as published by the Free Software Foundation; either version 2 of the License, or (at your option) any later version. This library is distributed in the hope that it will be useful, but WITHOUT ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU Lesser General Public License for more details. You should have received a copy of the GNU Lesser General Public License along with this library; if not, write to the Free Software Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA Also add information on how to contact you by electronic and paper mail. You should also get your employer (if you work as a programmer) or your school, if any, to sign a "copyright disclaimer" for the library, if necessary. Here is a sample; alter the names: Yoyodyne, Inc., hereby disclaims all copyright interest in the library `Frob' (a library for tweaking knobs) written by James Random Hacker. , 1 April 1990 Ty Coon, President of Vice That's all there is to it! usr/doc/sox-12.17.4/INSTALL0100644000000000000000000000662607614633276013635 0ustar rootrootSoX: Sound eXchange Installation This distribution will compile and run on most UNIX systems. It was originally developed on a UNIX/386 machine running AT&T V.3.2 but its currently developed under Linux. With little work it should work with most SVR4 systems, BSD-derived UNIX's and DOS/Windows systems that use the GNU tool set. Compiling --------- The preferred method for compiling SoX is to use the "configure" scripts compatible with most UNIX systems that contain "/bin/sh" or equivalent (even the Win95/NT Cygwin setup will work with this). To compile SoX on these platforms run the following commands: ./configure make make install You should then have a working version of SoX in the directory. There are several optional parameters that you may pass to the configure script to customize SoX for your applications. Run "./configure --help" for a complete list of options. If your system works with the "configure" script then you may skip to the Optional Compiling section. If your system does not work with the configure scripts then there are several canned "Makefile"'s that you can use. The following systems have a canned Makefile: DOS Makefile.dos (Borland or Turbo C) WIN95/NT Makefile.dos (Needs modifying for Visual C++) OS/2 Makefile.gcc (using EMX GCC compiler) To use a canned Makefile, a few steps need to be completed. First up, you need to make a copy of stconfig.h.in and call it stconfig.h. Modify this file to reflect your environment. After that, copy either Makefile.dos or Makefile.gcc to a file called Makefile. This needs to be done in both the main SoX directory and the ./gsm directory. This file also needs to be modified to reflect your environment. Optional Compile Features ------------------------- A GSM library is included with SoX. More information on this library can be obtained from http://www.cs.tu-berlin.de/~jutta/toast.html. If this library fails to compile on your system, you can specify --disable-gsm to prevent it from being compiled in. SoX can make use of Ogg Vorbis libraries to play and record Ogg Vorbis files. If this library is installed in a non-standard location in your system then you can use the CPPFLAGS and LDFLAGS variables to allow configure to find this library. For examle, "./configure CPPFLAGS=-I/home/sox/include LDFLAGS=-L/home/sox/lib". If you're not processing lots of u-law or A-law files and would like to save around 64K of memory when SoX is executed then you can use runtime routines to perform u-law/A-law conversions. This is slower then the default lookup tables but results in the same answers. To save this memory, specify --disable-fast-ulaw and --disable-fast-alaw. Testing ------- After successfully compiling SoX, try translating a sound file. If you can play one of the supported sound file formats, translate 'monkey.voc' to your format (we'll use 'xxx'): sox monkey.voc monkey.xxx You may have to give the word size and rate for the file. For example, this command will make a sound file with a data rate of 12,500 samples per second and the data formatted as signed shorts: sox monkey.voc -r 12500 -s -w monkey.xxx If monkey.xxx plays properly (it's a very short monkey screech), congratulations! SoX works. After that, running "tests.sh" and "testall.sh" ("tests.bat" and "testall.bat" for DOS) tests most of the implemented file handlers to make sure that some portability issue haven't popped up. usr/doc/sox-12.17.4/README0100644000000000000000000001016607637402016013445 0ustar rootrootSoX: Sound eXchange ------------------- SoX (also known as Sound eXchange) translates sound files between different file formats, and optionally applies various sound effects. SoX is intended as the Swiss Army knife of sound processing tools. It doesn't do anything very well, but sooner or later it comes in very handy. This release understands: o Raw files in various binary formats o Raw textual data o Amiga 8svx files o Apple/SGI AIFF files o SUN .au style files o PCM, u-law, A-law o G7xx ADPCM files (read only) o mutant DEC .au files o NeXT .snd files o AVR files o CD-R (music CD format) o CVS and VMS files (continuous variable slope) o GSM files o Macintosh HCOM files o Amiga MAUD files o MP3 files (with optional external library) 0 Psion Record.app files o IRCAM SoundFile files o NIST SPHERE files o Turtle beach SampleVision files. o Soundtool (DOS) files o Yamaha TX-16W sampler files. o Sound Blaster .VOC files o Ogg Vorbis files (with optional external library) o Microsoft .WAV files o PCM, u-law, A-law o MS ADPCM, IMA ADPCM o GSM o Psion (palmtop) A-law WVE files o Pseudo-file formats that allow direct playing/recording from some audio devices under UNIX. o Pseudo-nul file that reads and writes from/to nowhere The sound effects include: o Channel averaging, duplication, and removal o Band-pass filters o Band-reject filter o Chorus effect o DCShift audio. Useful to get the best volume adjustments. o Deemphases filter o Move sound stage of CD audio to in front of you (for headphone use) o Add an echo o Add a sequence of echos o Fade in and out o Apply a flanger effect o Apply a high-pass filter o Apply a low-pass filter o Display a list of loops in a file o Add masking noise to a signal o Pan sound between channels o Apply a phaser effect o Change the pitch of a sound file without effecting its speed o Change sampling rates using several different algorithms. A 'resample' and 'polyphase' effect use high-grade signal rate changes using real signal theory! o Apply a reverb effect o Reverse the sound samples (to search for Satanic messages ;-) o Detect periods of silence and start and stop processing based on it o Change the speed of samples being played (like speeding up the motor on a tape recorder) o Display general stats on a sound sample o Stretch/shorten the duration of a sound file (without effecting pitch). o Swap stereo channels o Create sounds with a simple synthesizer. o Trim audio data from beginning and end of file. o Add the world-famous Fender Vibro-Champ effect o Adjust volume of samples History: This is release 12.17.3 of SoX and its companion library Sound Tools. SoX was originally written and maintained by Lance Norskog but is currently maintained by Chris Bagwell (cbagwell@users.sourceforge.net). Installing: Unless your using a pre-compiled binary version, you will need to compile SoX as described in the INSTALL file. Please read that file for further instructions. SoX is distributed with two text files named sox.txt and soxexam.txt. These files give a background on how SoX deals with sound files and what command line options are available to convert a sound file to another file format and how to apply sound effects. SoX has an auto-detect feature that attempts to figure out the nature of a sound file. It does this by looking for 'magic values' in the header of the audio file. If it is unable to detect the format of the sound file then the user must use command line options to inform SoX. I hope to inspire the creation of a common base of sound processing tools for computer multimedia work, similar to the PBM toolkit for image manipulation. If you have bug fixes/enhancements, please send it to me as I would like to coordinate the releases. Please document your changes. I do not possess every kind of computer currently sold, and SoX is now beyond the phase where I can understand and test most of your contributions. The majority of SoX features and source code are contributed by you the user. Thank you very much for making SoX a success! usr/doc/sox-12.17.4/TODO0100644000000000000000000000446007503250346013254 0ustar rootrootPeople are encouraged to pick some of these and implement it. Send all patches to cbagwell@users.sourceforge.net. o Add new PRC format in to auto.c o Still passing some options using CFLAGS instead of stconfig.h. o Lots of effects are using "double" to prevent overflow of st_sample_t math. With ststdint.h we now have a portable way of detecting uin64_t... Use that when possible to speed up math (won't have to convert back and forth from int to float). o Several files are using hard-coded #'s instead of ST_SAMPLE_MIN/MAX. Change these. o Make a global version of MIN/MAX instead of sprinkled min/max/MIN/MAX o Make a global version of clip() instead of sprinked in all files. o Add support to all file handlers to handle 32-bit and float data types since raw functions can handle them. o All comment code has a memory leak. They must malloc memory to store comment but its never free()'d. o Comment strings. Some file formats have space for embedded comments. These are currently thrown away. Printing them out, carrying them forward, and an option to add new ones would be handy. o Add support for .TXT format as Cooledit supports. Not really fit for graphing since it is only a stream of ascii sample values but some program appear to like it. Also has a much nicer header. http://lab.chass.utoronto.ca/damion/audio/coolwav.htm o Keep sox from using "fail" on errors. Sox was supposed to be a sound library called "ST" but libraries shouldn't exit a program, they should return error codes for users to handle. Initial support for this has been added. Needs to be completed. o Enhance general robustness... For instance, malloc is called in lots of places without checking its return value. See last option as well. SOX includes skeleton format files to assist you in supporting new formats, sound effect loops, and special-purpose programs. The full skeleton format, skel.c, helps you write a driver for a new format which has data structures. Skeleff.c is a starting point for writing a sound effect loop. Sox.c is a good starting point for new programs. (Someone finally did this and told me what was wrong...) 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