ref: 100706dd3b9ddc73fc46fc3da414d50a17a11f83
parent: 9c8953be8cd2734442ad3c3811c94bfaf81d57f4
author: cbagwell <cbagwell>
date: Wed Sep 8 20:52:11 EDT 2004
-v is now a format option which allows max_input_files to increase to 32.
--- a/Changelog
+++ b/Changelog
@@ -27,6 +27,11 @@
cause SoX to stick in a loop forever. Now, it will
abort on IDv3 tags larger then 100k. Could still be
improved to handle any size.
+ o Changed volume option (-v) so that it tracks the file
+ it was specified. This means that when specified with
+ the input file, it changes volume before effects engine
+ and when specified with output file, its done after effects
+ engine.
sox-12.17.5
-----------
--- a/sox.1
+++ b/sox.1
@@ -14,20 +14,22 @@
sox \- Sound eXchange : universal sound sample translator
.SH SYNOPSIS
.P
-\fBsox\fR \fIinfile outfile\fR
+\fBsox\fR \fIinfile1\fR [ \fIinfile2\fR ... ] \fIoutfile\fR
.P
-\fBsox\fR [ \fIgeneral options\fR ] [ \fIformat options\fR ] \fIinfile\fR
+\fBsox\fR [ \fIgeneral options\fR ] [ \fIformat options\fR ] \fIinfile1\fR
.br
- [ \fIformat options\fR ] \fIoutfile\fR
+ [ [ \fIformat options\fR ] \fIinfile2\fR ... ] [ \fIformat options\fR ] \fIoutfile\fR
.br
[ \fIeffect\fR [ \fIeffect options\fR ] ... ]
.P
-\fBsoxmix\fR \fIinfile1 infile2 outfile\fR
+\fBsoxmix\fR \fIinfile1 infile2\fR [ \fIinfile3\fR ... ] outfile\fR
.P
\fBsoxmix\fR [ \fIgeneral options\fR ] [ \fIformat options\fR ] \fIinfile1\fR
.br
[ \fIformat options\fR ] \fIinfile2\fR
.br
+ [ [ \fIformat options\fR ] \fIinfile3\fR ... ]
+.br
[ \fIformat options\fR ] \fIoutfile\fR
.br
[ \fIeffect\fR [ \fIeffect options\fR ] ... ]
@@ -35,12 +37,12 @@
.P
.B General options:
.br
- [ -h ] [ -p ] [ -v \fIvolume\fR ] [ -V ]
+ [ -h ] [ -p ] [ -V ]
.P
.B Format options:
.br
[ -t \fIfiletype\fR ] [ -r \fIrate\fR ] [ -s/-u/-U/-A/-a/-i/-g/-f ]
- [ -b/-w/-l/-d ]
+ [ -b/-w/-l/-d ] [ -v \fIvolume\fR ]
[ -c \fIchannels\fR ] [ -x ] [ -e ]
.P
.B Effects:
@@ -139,16 +141,20 @@
is a command line program that can convert most popular audio files
to most other popular audio file formats. It can optionally change
the audio sample data type and apply one or more
-sound effects to the file during this translation.
+sound effects to the file during this translation.
.P
+If more then one input file is specified then they are concatenated into the
+output file. In this case, it has a restriction that all input files
+must be of the same data type and sample rates.
+.P
.I soxmix
is functionally the same as the command line program
.I sox
-expect that it takes two files as input and mixes the audio together
-to produce a single file as output. It has a restriction that both
+expect that it takes two or more files as input and mixes the audio together
+to produce a single file as output. It has a restriction that all
input files must be of the same data type and sample rates.
.P
-There are two types of audio files formats that
+There are two types of audio file formats that
.I SoX
can work with. The first are self-describing file formats. These
contain a header that completely describe the characteristics of
@@ -227,6 +233,32 @@
sample rate changing effect is not specified then a default one will internally
be ran by SoX using its default parameters.
.TP 10
+\fB-v \fIvolume\fR
+Change amplitude (floating point);
+less than 1.0 decreases, greater than 1.0 increases. May use a negative
+number to invert the phase of the audio data. It is interesting to note
+that we perceive volume
+logarithmically but this adjusts the amplitude linearly.
+.br
+As with other format options, the volume option effects the
+file its specified with. This is useful whe processing mutiple
+input files as the volume adjustment can be specified for each
+input file or just once to adjust the output file. This can be
+compared to an audio mixer were you can control the volume of
+each input as well as a master volume (output side).
+.br
+\fIsoxmix\fR defaults the value of the -v option for each input
+file to 1/input_file_count. This means if your mixing two
+input files together then each input file's volume is adjusted
+by 0.5. This is done to prevent clipping of audio data during
+the mixing operation.
+Users will most likely not be happy with this large of a volume adjustment
+and can specify the -v option to override this default value.
+.br
+Note: For the non-mixing case, see the \fBstat\fR effect for information on
+finding the maximum volume adjustment that can be done with this option
+without causing audio data to be clipped.
+.TP 10
\fB-s/-u/-U/-A/-a/-i/-g/-f\fR
The sample data encoding is signed linear (2's complement),
unsigned linear, u-law (logarithmic), A-law (logarithmic),
@@ -299,17 +331,6 @@
a different rate than the input file. Currently, this defaults to
using the \fBrate\fR effect instead of the \fBresample\fR effect for sample
rate changes.
-.TP 10
-\fB-v \fIvolume\fR
-Change amplitude (floating point);
-less than 1.0 decreases, greater than 1.0 increases. May use a negative
-number to invert the phase of the audio data. It is interesting to note
-that we perceive volume
-logarithmically but this adjusts the amplitude linearly.
-.br
-Note: see the \fBstat\fR effect for information on finding the maximum
-value that can be used with this option without causing audio data be
-be clipped.
.TP 10
\fB-V\fR
Print a description of processing phases.
--- a/sox.txt
+++ b/sox.txt
@@ -6,26 +6,27 @@
sox - Sound eXchange : universal sound sample translator
SYNOPSIS
- sox infile outfile
+ sox infile1 [ infile2 ... ] outfile
- sox [ general options ] [ format options ] infile
- [ format options ] outfile
+ sox [ general options ] [ format options ] infile1
+ [ [ format options ] infile2 ... ] [ format options ] outfile
[ effect [ effect options ] ... ]
- soxmix infile1 infile2 outfile
+ soxmix infile1 infile2 [ infile3 ... ] outfile
soxmix [ general options ] [ format options ] infile1
[ format options ] infile2
+ [ [ format options ] infile3 ... ]
[ format options ] outfile
[ effect [ effect options ] ... ]
General options:
- [ -h ] [ -p ] [ -v volume ] [ -V ]
+ [ -h ] [ -p ] [ -V ]
Format options:
[ -t filetype ] [ -r rate ] [ -s/-u/-U/-A/-a/-i/-g/-f ]
- [ -b/-w/-l/-d ]
+ [ -b/-w/-l/-d ] [ -v volume ]
[ -c channels ] [ -x ] [ -e ]
Effects:
@@ -52,7 +53,6 @@
highpass frequency
lowp frequency
lowpass frequency
- map
mask
pan direction
phaser gain-in gain-out delay decay speed < -s | -t >
@@ -85,12 +85,16 @@
audio sample data type and apply one or more sound effects to the file
during this translation.
+ If more then one input file is specified then they are concatenated
+ into the output file. In this case, it has a restriction that all
+ input files must be of the same data type and sample rates.
+
soxmix is functionally the same as the command line program sox expect
- that it takes two files as input and mixes the audio together to pro-
- duce a single file as output. It has a restriction that both input
- files must be of the same data type and sample rates.
+ that it takes two or more files as input and mixes the audio together
+ to produce a single file as output. It has a restriction that all
+ input files must be of the same data type and sample rates.
- There are two types of audio files formats that SoX can work with. The
+ There are two types of audio file formats that SoX can work with. The
first are self-describing file formats. These contain a header that
completely describe the characteristics of the audio data that follows.
@@ -158,63 +162,87 @@
changing effect is not specified then a default one will
internally be ran by SoX using its default parameters.
+ -v volume Change amplitude (floating point); less than 1.0 decreases,
+ greater than 1.0 increases. May use a negative number to
+ invert the phase of the audio data. It is interesting to
+ note that we perceive volume logarithmically but this adjusts
+ the amplitude linearly.
+ As with other format options, the volume option effects the
+ file its specified with. This is useful whe processing muti-
+ ple input files as the volume adjustment can be specified for
+ each input file or just once to adjust the output file. This
+ can be compared to an audio mixer were you can control the
+ volume of each input as well as a master volume (output
+ side).
+ soxmix defaults the value of the -v option for each input
+ file to 1/input_file_count. This means if your mixing two
+ input files together then each input file’s volume is
+ adjusted by 0.5. This is done to prevent clipping of audio
+ data during the mixing operation. Users will most likely not
+ be happy with this large of a volume adjustment and can spec-
+ ify the -v option to override this default value.
+ Note: For the non-mixing case, see the stat effect for infor-
+ mation on finding the maximum volume adjustment that can be
+ done with this option without causing audio data to be
+ clipped.
+
-s/-u/-U/-A/-a/-i/-g/-f
- The sample data encoding is signed linear (2’s complement),
- unsigned linear, u-law (logarithmic), A-law (logarithmic),
+ The sample data encoding is signed linear (2’s complement),
+ unsigned linear, u-law (logarithmic), A-law (logarithmic),
ADPCM, IMA_ADPCM, GSM, or Floating-point.
- U-law (actually shorthand for mu-law) and A-law are the U.S.
- and international standards for logarithmic telephone sound
- compression. When uncompressed u-law has roughly the preci-
- sion of 14-bit PCM audio and A-law has roughly the precision
+ U-law (actually shorthand for mu-law) and A-law are the U.S.
+ and international standards for logarithmic telephone sound
+ compression. When uncompressed u-law has roughly the preci-
+ sion of 14-bit PCM audio and A-law has roughly the precision
of 13-bit PCM audio.
- A-law and u-law data is sometimes encoded using a reversed
- bit-ordering (ie. MSB becomes LSB). Internally, SoX under-
- stands how to work with this encoding but there is currently
- no command line option to specify it. If you need this sup-
- port then you can use the psuedo file types of ".la" and
- ".lu" to inform sox of the encoding. See supported file
+ A-law and u-law data is sometimes encoded using a reversed
+ bit-ordering (ie. MSB becomes LSB). Internally, SoX under-
+ stands how to work with this encoding but there is currently
+ no command line option to specify it. If you need this sup-
+ port then you can use the psuedo file types of ".la" and
+ ".lu" to inform sox of the encoding. See supported file
types for more information.
- ADPCM is a form of sound compression that has a good compro-
- mise between good sound quality and fast encoding/decoding
- time. It is used for telephone sound compression and places
+ ADPCM is a form of sound compression that has a good compro-
+ mise between good sound quality and fast encoding/decoding
+ time. It is used for telephone sound compression and places
were full fidelity is not as important. When uncompressed it
- has roughly the precision of 16-bit PCM audio. Popular ver-
+ has roughly the precision of 16-bit PCM audio. Popular ver-
sion of ADPCM include G.726, MS ADPCM, and IMA ADPCM. The -a
- flag has different meanings in different file handlers. In
- .wav files it represents MS ADPCM files, in all others it
- means G.726 ADPCM. IMA ADPCM is a specific form of ADPCM
- compression, slightly simpler and slightly lower fidelity
- than Microsoft’s flavor of ADPCM. IMA ADPCM is also called
+ flag has different meanings in different file handlers. In
+ .wav files it represents MS ADPCM files, in all others it
+ means G.726 ADPCM. IMA ADPCM is a specific form of ADPCM
+ compression, slightly simpler and slightly lower fidelity
+ than Microsoft’s flavor of ADPCM. IMA ADPCM is also called
DVI ADPCM.
- GSM is a standard used for telephone sound compression in
- European countries and its gaining popularity because of its
- quality. It usually is CPU intensive to work with GSM audio
+ GSM is a standard used for telephone sound compression in
+ European countries and its gaining popularity because of its
+ quality. It usually is CPU intensive to work with GSM audio
data.
-b/-w/-l/-d
- The sample data size is in bytes, 16-bit words, 32-bit long
+ The sample data size is in bytes, 16-bit words, 32-bit long
words, or 64-bit double long (long long) words.
- -x The sample data is in XINU format; that is, it comes from a
- machine with the opposite word order than yours and must be
- swapped according to the word-size given above. Only 16-bit
- and 32-bit integer data may be swapped. Machine-format
+ -x The sample data is in XINU format; that is, it comes from a
+ machine with the opposite word order than yours and must be
+ swapped according to the word-size given above. Only 16-bit
+ and 32-bit integer data may be swapped. Machine-format
floating-point data is not portable.
-c channels
- The number of sound channels in the data file. This may be
- 1, 2, or 4; for mono, stereo, or quad sound data. To cause
- the output file to have a different number of channels than
- the input file, include this option with the output file
- options. If the input and output file have a different num-
+ The number of sound channels in the data file. This may be
+ 1, 2, or 4; for mono, stereo, or quad sound data. To cause
+ the output file to have a different number of channels than
+ the input file, include this option with the output file
+ options. If the input and output file have a different num-
ber of channels then the avg effect must be used. If the avg
- effect is not specified on the command line it will be
+ effect is not specified on the command line it will be
invoked internally with default parameters.
-e When used after the input filename (so that it applies to the
output file) it allows you to avoid giving an output filename
and will not produce an output file. It will apply any spec-
- ified effects to the input file. This is mainly useful with
+ ified effects to the input file. This is mainly useful with
the stat effect but can be used with others.
General options:
@@ -221,35 +249,26 @@
-h Print version number and usage information.
- -p Run in preview mode and run fast. This will somewhat speed
+ -p Run in preview mode and run fast. This will somewhat speed
up SoX when the output format has a different number of chan-
- nels and a different rate than the input file. Currently,
- this defaults to using the rate effect instead of the resam-
+ nels and a different rate than the input file. Currently,
+ this defaults to using the rate effect instead of the resam-
ple effect for sample rate changes.
- -v volume Change amplitude (floating point); less than 1.0 decreases,
- greater than 1.0 increases. May use a negative number to
- invert the phase of the audio data. It is interesting to
- note that we perceive volume logarithmically but this adjusts
- the amplitude linearly.
- Note: see the stat effect for information on finding the max-
- imum value that can be used with this option without causing
- audio data be be clipped.
-
- -V Print a description of processing phases. Useful for figur-
+ -V Print a description of processing phases. Useful for figur-
ing out exactly how SoX is mangling your sound samples.
FILE TYPES
SoX attempts to determine the file type of input files automatically by
- looking at the header of the audio file. When it is unable to detect
- the file type or if its an output file then it uses the file extension
+ looking at the header of the audio file. When it is unable to detect
+ the file type or if its an output file then it uses the file extension
of the file to determine what type of file format handler to use. This
can be overridden by specifying the "-t" option on the command line.
- The input and output files may be read from standard in and out. This
+ The input and output files may be read from standard in and out. This
is done by specifying ’-’ as the filename.
- File formats which have headers are checked, if that header doesn’t
+ File formats which have headers are checked, if that header doesn’t
seem right, the program exits with an appropriate message.
The following file formats are supported:
@@ -257,19 +276,19 @@
.8svx Amiga 8SVX musical instrument description format.
- .aiff AIFF files used on Apple IIc/IIgs and SGI. Note: the AIFF
- format supports only one SSND chunk. It does not support
- multiple sound chunks, or the 8SVX musical instrument
- description format. AIFF files are multimedia archives and
- can have multiple audio and picture chunks. You may need a
+ .aiff AIFF files used on Apple IIc/IIgs and SGI. Note: the AIFF
+ format supports only one SSND chunk. It does not support
+ multiple sound chunks, or the 8SVX musical instrument
+ description format. AIFF files are multimedia archives and
+ can have multiple audio and picture chunks. You may need a
separate archiver to work with them.
- .au SUN Microsystems AU files. There are apparently many types
+ .au SUN Microsystems AU files. There are apparently many types
of .au files; DEC has invented its own with a different magic
- number and word order. The .au handler can read these files
- but will not write them. Some .au files have valid AU head-
+ number and word order. The .au handler can read these files
+ but will not write them. Some .au files have valid AU head-
ers and some do not. The latter are probably original SUN u-
- law 8000 hz samples. These can be dealt with using the .ul
+ law 8000 hz samples. These can be dealt with using the .ul
format (see below).
.avr Audio Visual Research
@@ -278,56 +297,56 @@
.cdr CD-R
CD-R files are used in mastering music on Compact Disks. The
- audio data on a CD-R disk is a raw audio file with a format
- of stereo 16-bit signed samples at a 44khz sample rate.
- There is a special blocking/padding oddity at the end of the
+ audio data on a CD-R disk is a raw audio file with a format
+ of stereo 16-bit signed samples at a 44khz sample rate.
+ There is a special blocking/padding oddity at the end of the
audio file and is why it needs its own handler.
.cvs Continuously Variable Slope Delta modulation
- Used to compress speech audio for applications such as voice
+ Used to compress speech audio for applications such as voice
mail.
.dat Text Data files
- These files contain a textual representation of the sample
- data. There is one line at the beginning that contains the
- sample rate. Subsequent lines contain two numeric data
- items: the time since the beginning of the first sample and
- the sample value. Values are normalized so that the maximum
+ These files contain a textual representation of the sample
+ data. There is one line at the beginning that contains the
+ sample rate. Subsequent lines contain two numeric data
+ items: the time since the beginning of the first sample and
+ the sample value. Values are normalized so that the maximum
and minimum are 1.00 and -1.00. This file format can be used
- to create data files for external programs such as FFT ana-
- lyzers or graph routines. SoX can also convert a file in
+ to create data files for external programs such as FFT ana-
+ lyzers or graph routines. SoX can also convert a file in
this format back into one of the other file formats.
.gsm GSM 06.10 Lossy Speech Compression
A standard for compressing speech which is used in the Global
- Standard for Mobil telecommunications (GSM). Its good for
+ Standard for Mobil telecommunications (GSM). Its good for
its purpose, shrinking audio data size, but it will introduce
- lots of noise when a given sound sample is encoded and
- decoded multiple times. This format is used by some voice
+ lots of noise when a given sound sample is encoded and
+ decoded multiple times. This format is used by some voice
mail applications. It is rather CPU intensive.
GSM in SoX is optional and requires access to an external GSM
- library. To see if there is support for gsm run sox -h and
+ library. To see if there is support for gsm run sox -h and
look for it under the list of supported file formats.
- .hcom Macintosh HCOM files. These are (apparently) Mac FSSD files
- with some variant of Huffman compression. The Macintosh has
+ .hcom Macintosh HCOM files. These are (apparently) Mac FSSD files
+ with some variant of Huffman compression. The Macintosh has
wacky file formats and this format handler apparently doesn’t
- handle all the ones it should. Mac users will need your
- usual arsenal of file converters to deal with an HCOM file
+ handle all the ones it should. Mac users will need your
+ usual arsenal of file converters to deal with an HCOM file
under Unix or DOS.
.maud An Amiga format
- An IFF-conform sound file type, registered by MS MacroSystem
- Computer GmbH, published along with the "Toccata" sound-card
+ An IFF-conform sound file type, registered by MS MacroSystem
+ Computer GmbH, published along with the "Toccata" sound-card
on the Amiga. Allows 8bit linear, 16bit linear, A-Law, u-law
in mono and stereo.
.mp3 MP3 Compressed Audio
- MP3 audio files come from the MPEG standards for audio and
- video compression. They are a lossy compression format that
- achieves good compression rates with a minimum amount of
+ MP3 audio files come from the MPEG standards for audio and
+ video compression. They are a lossy compression format that
+ achieves good compression rates with a minimum amount of
quality loss. Also see Ogg Vorbis for a similar format. MP3
- support in SoX is optional and requires access to either or
+ support in SoX is optional and requires access to either or
both the external libmad and libmp3lame libraries. To see if
there is support for Mp3 run sox -h and look for it under the
list of supported file formats as "mp3".
@@ -335,64 +354,64 @@
.nul Null file handler. This is a fake file hander that act as if
its reading a stream of 0’s from a while or fake writing out-
- put to a file. This is not a very useful file handler in
- most cases. It might be useful in some scripts were you do
- not want to read or write from a real file but would like to
+ put to a file. This is not a very useful file handler in
+ most cases. It might be useful in some scripts were you do
+ not want to read or write from a real file but would like to
specify a filename for consistency.
.ogg Ogg Vorbis Compressed Audio.
- Ogg Vorbis is a open, patent-free CODEC designed for com-
- pressing music and streaming audio. It is similar to MP3,
- VQF, AAC, and other lossy formats. SoX can decode all types
+ Ogg Vorbis is a open, patent-free CODEC designed for com-
+ pressing music and streaming audio. It is similar to MP3,
+ VQF, AAC, and other lossy formats. SoX can decode all types
of Ogg Vorbis files, but can only encode at 128 kbps. Decod-
ing is somewhat CPU intensive and encoding is very CPU inten-
sive.
Ogg Vorbis in SoX is optional and requires access to external
- Ogg Vorbis libraries. To see if there is support for Ogg
+ Ogg Vorbis libraries. To see if there is support for Ogg
Vorbis run sox -h and look for it under the list of supported
file formats as "vorbis".
ossdsp OSS /dev/dsp device driver
- This is a pseudo-file type and can be optionally compiled
- into SoX. Run sox -h to see if you have support for this
+ This is a pseudo-file type and can be optionally compiled
+ into SoX. Run sox -h to see if you have support for this
file type. When this driver is used it allows you to open up
- the OSS /dev/dsp file and configure it to use the same data
- format as passed in to SoX. It works for both playing and
- recording sound samples. When playing sound files it
- attempts to set up the OSS driver to use the same format as
- the input file. It is suggested to always override the out-
+ the OSS /dev/dsp file and configure it to use the same data
+ format as passed in to SoX. It works for both playing and
+ recording sound samples. When playing sound files it
+ attempts to set up the OSS driver to use the same format as
+ the input file. It is suggested to always override the out-
put values to use the highest quality samples your sound card
can handle. Example: -t ossdsp -w -s /dev/dsp
.prc Psion record.app
Used in some Psion devices for System alarms. This format is
- newer then the .wve format that is used in some Psion
+ newer then the .wve format that is used in some Psion
devices.
.sf IRCAM Sound Files.
- Sound Files are used by academic music software such as the
+ Sound Files are used by academic music software such as the
CSound package, and the MixView sound sample editor.
.sph
- SPHERE (SPeech HEader Resources) is a file format defined by
- NIST (National Institute of Standards and Technology) and is
- used with speech audio. SoX can read these files when they
+ SPHERE (SPeech HEader Resources) is a file format defined by
+ NIST (National Institute of Standards and Technology) and is
+ used with speech audio. SoX can read these files when they
contain u-law and PCM data. It will ignore any header infor-
- mation that says the data is compressed using shorten com-
- pression and will treat the data as either u-law or PCM.
- This will allow SoX and the command line shorten program to
- be ran together using pipes to uncompress the data and then
+ mation that says the data is compressed using shorten com-
+ pression and will treat the data as either u-law or PCM.
+ This will allow SoX and the command line shorten program to
+ be ran together using pipes to uncompress the data and then
pass the result to SoX for processing.
.smp Turtle Beach SampleVision files.
SMP files are for use with the PC-DOS package SampleVision by
- Turtle Beach Softworks. This package is for communication to
- several MIDI samplers. All sample rates are supported by the
+ Turtle Beach Softworks. This package is for communication to
+ several MIDI samplers. All sample rates are supported by the
package, although not all are supported by the samplers them-
selves. Currently loop points are ignored.
.snd
- Under DOS this file format is the same as the .sndt format.
+ Under DOS this file format is the same as the .sndt format.
Under all other platforms it is the same as the .au format.
.sndt SoundTool files.
@@ -399,117 +418,117 @@
This is an older DOS file format.
sunau Sun /dev/audio device driver
- This is a pseudo-file type and can be optionally compiled
- into SoX. Run sox -h to see if you have support for this
+ This is a pseudo-file type and can be optionally compiled
+ into SoX. Run sox -h to see if you have support for this
file type. When this driver is used it allows you to open up
- a Sun /dev/audio file and configure it to use the same data
- type as passed in to SoX. It works for both playing and
- recording sound samples. When playing sound files it
+ a Sun /dev/audio file and configure it to use the same data
+ type as passed in to SoX. It works for both playing and
+ recording sound samples. When playing sound files it
attempts to set up the audio driver to use the same format as
- the input file. It is suggested to always override the out-
- put values to use the highest quality samples your hardware
- can handle. Example: -t sunau -w -s /dev/audio or -t sunau
+ the input file. It is suggested to always override the out-
+ put values to use the highest quality samples your hardware
+ can handle. Example: -t sunau -w -s /dev/audio or -t sunau
-U -c 1 /dev/audio for older sun equipment.
.txw Yamaha TX-16W sampler.
- A file format from a Yamaha sampling keyboard which wrote
- IBM-PC format 3.5" floppies. Handles reading of files which
- do not have the sample rate field set to one of the expected
- by looking at some other bytes in the attack/loop length
- fields, and defaulting to 33kHz if the sample rate is still
+ A file format from a Yamaha sampling keyboard which wrote
+ IBM-PC format 3.5" floppies. Handles reading of files which
+ do not have the sample rate field set to one of the expected
+ by looking at some other bytes in the attack/loop length
+ fields, and defaulting to 33kHz if the sample rate is still
unknown.
.vms More info to come.
- Used to compress speech audio for applications such as voice
+ Used to compress speech audio for applications such as voice
mail.
.voc Sound Blaster VOC files.
- VOC files are multi-part and contain silence parts, looping,
- and different sample rates for different chunks. On input,
- the silence parts are filled out, loops are rejected, and
+ VOC files are multi-part and contain silence parts, looping,
+ and different sample rates for different chunks. On input,
+ the silence parts are filled out, loops are rejected, and
sample data with a new sample rate is rejected. Silence with
- a different sample rate is generated appropriately. On out-
- put, silence is not detected, nor are impossible sample
- rates. Note, this version now supports playing VOC files
+ a different sample rate is generated appropriately. On out-
+ put, silence is not detected, nor are impossible sample
+ rates. Note, this version now supports playing VOC files
with multiple blocks and supports playing files containing u-
law and A-law samples.
vorbis See .ogg format.
- vox A headerless file of Dialogic/OKI ADPCM audio data commonly
- comes with the extension .vox. This ADPCM data has 12-bit
+ vox A headerless file of Dialogic/OKI ADPCM audio data commonly
+ comes with the extension .vox. This ADPCM data has 12-bit
precision packed into only 4-bits.
.wav Microsoft .WAV RIFF files.
- These appear to be very similar to IFF files, but not the
- same. They are the native sound file format of Windows.
- (Obviously, Windows was of such incredible importance to the
+ These appear to be very similar to IFF files, but not the
+ same. They are the native sound file format of Windows.
+ (Obviously, Windows was of such incredible importance to the
computer industry that it just had to have its own sound file
format.) Normally .wav files have all formatting information
in their headers, and so do not need any format options spec-
- ified for an input file. If any are, they will override the
- file header, and you will be warned to this effect. You had
- better know what you are doing! Output format options will
- cause a format conversion, and the .wav will written appro-
- priately. SoX currently can read PCM, ULAW, ALAW, MS ADPCM,
- and IMA (or DVI) ADPCM. It can write all of these formats
+ ified for an input file. If any are, they will override the
+ file header, and you will be warned to this effect. You had
+ better know what you are doing! Output format options will
+ cause a format conversion, and the .wav will written appro-
+ priately. SoX currently can read PCM, ULAW, ALAW, MS ADPCM,
+ and IMA (or DVI) ADPCM. It can write all of these formats
including (NEW!) the ADPCM encoding.
.wve Psion 8-bit A-law
- These are 8-bit A-law 8khz sound files used on the Psion
+ These are 8-bit A-law 8khz sound files used on the Psion
palmtop portable computer.
.raw Raw files (no header).
The sample rate, size (byte, word, etc), and encoding
- (signed, unsigned, etc.) of the sample file must be given.
+ (signed, unsigned, etc.) of the sample file must be given.
The number of channels defaults to 1.
.ub, .sb, .uw, .sw, .ul, .al, .lu, .la, .sl
These are several suffices which serve as a shorthand for raw
- files with a given size and encoding. Thus, ub, sb, uw, sw,
- ul, al, lu, la and sl correspond to "unsigned byte", "signed
- byte", "unsigned word", "signed word", "u-law" (byte), "A-
+ files with a given size and encoding. Thus, ub, sb, uw, sw,
+ ul, al, lu, la and sl correspond to "unsigned byte", "signed
+ byte", "unsigned word", "signed word", "u-law" (byte), "A-
law" (byte), inverse bit order "u-law", inverse bit order "A-
law", and "signed long". The sample rate defaults to 8000 hz
if not explicitly set, and the number of channels defaults to
- 1. There are lots of Sparc samples floating around in u-law
- format with no header and fixed at a sample rate of 8000 hz.
- (Certain sound management software cheerfully ignores the
- headers.) Similarly, most Mac sound files are in unsigned
+ 1. There are lots of Sparc samples floating around in u-law
+ format with no header and fixed at a sample rate of 8000 hz.
+ (Certain sound management software cheerfully ignores the
+ headers.) Similarly, most Mac sound files are in unsigned
byte format with a sample rate of 11025 or 22050 hz.
- .auto This is a ‘‘meta-type’’: specifying this type for an input
- file triggers some code that tries to guess the real type by
- looking for magic words in the header. If the type can’t be
- guessed, the program exits with an error message. The input
- must be a plain file, not a pipe. This type can’t be used
+ .auto This is a ‘‘meta-type’’: specifying this type for an input
+ file triggers some code that tries to guess the real type by
+ looking for magic words in the header. If the type can’t be
+ guessed, the program exits with an error message. The input
+ must be a plain file, not a pipe. This type can’t be used
for output files.
EFFECTS
- Multiple effects may be applied to the audio data by specifying them
+ Multiple effects may be applied to the audio data by specifying them
one after another at the end of the command line.
avg [ -l | -r | -f | -b | -1 | -2 | -3 | -4 | n,n,...,n ]
- Reduce the number of channels by averaging the samples, or
- duplicate channels to increase the number of channels. This
- effect is automatically used when the number of input chan-
- nels differ from the number of output channels. When reduc-
+ Reduce the number of channels by averaging the samples, or
+ duplicate channels to increase the number of channels. This
+ effect is automatically used when the number of input chan-
+ nels differ from the number of output channels. When reduc-
ing the number of channels it is possible to manually specify
- the avg effect and use the -l, -r, -f, -b, -1, -2, -3, -4,
- options to select only the left, right, front, back chan-
- nel(s) or specific channel for the output instead of averag-
- ing the channels. The -l, and -r options will do averaging
- in quad-channel files so select the exact channel to prevent
+ the avg effect and use the -l, -r, -f, -b, -1, -2, -3, -4,
+ options to select only the left, right, front, back chan-
+ nel(s) or specific channel for the output instead of averag-
+ ing the channels. The -l, and -r options will do averaging
+ in quad-channel files so select the exact channel to prevent
this.
- The avg effect can also be invoked with up to 16 double-pre-
- cision numbers, which specify the proportion (0.0 = 0% and
- 1.0 = 100%) of each input channel that is to be mixed into
- each output channel. In two-channel mode, 4 numbers are
- given: l->l, l->r, r->l, and r->r, respectively. In four-
- channel mode, the first 4 numbers give the proportions for
- the left-front output channel, as follows: lf->lf, rf->lf,
- lb->lf, and rb->rf. The next 4 give the right-front output
+ The avg effect can also be invoked with up to 16 double-pre-
+ cision numbers, which specify the proportion (0.0 = 0% and
+ 1.0 = 100%) of each input channel that is to be mixed into
+ each output channel. In two-channel mode, 4 numbers are
+ given: l->l, l->r, r->l, and r->r, respectively. In four-
+ channel mode, the first 4 numbers give the proportions for
+ the left-front output channel, as follows: lf->lf, rf->lf,
+ lb->lf, and rb->rf. The next 4 give the right-front output
in the same order, then left-back and right-back.
It is also possible to use the 16 numbers to expand or reduce
@@ -532,15 +551,15 @@
band [ -n ] center [ width ]
Apply a band-pass filter. The frequency response drops loga-
rithmically around the center frequency. The width gives the
- slope of the drop. The frequencies at center + width and
- center - width will be half of their original amplitudes.
- Band defaults to a mode oriented to pitched signals, i.e.
- voice, singing, or instrumental music. The -n (for noise)
+ slope of the drop. The frequencies at center + width and
+ center - width will be half of their original amplitudes.
+ Band defaults to a mode oriented to pitched signals, i.e.
+ voice, singing, or instrumental music. The -n (for noise)
option uses the alternate mode for un-pitched signals. Warn-
- ing: -n introduces a power-gain of about 11dB in the filter,
- so beware of output clipping. Band introduces noise in the
+ ing: -n introduces a power-gain of about 11dB in the filter,
+ so beware of output clipping. Band introduces noise in the
shape of the filter, i.e. peaking at the center frequency and
- settling around it. See filter for a bandpass effect with
+ settling around it. See filter for a bandpass effect with
steeper shoulders.
bandpass frequency bandwidth
@@ -552,11 +571,11 @@
chorus gain-in gain-out delay decay speed depth
-s | -t [ delay decay speed depth -s | -t ... ]
- Add a chorus to a sound sample. Each quadtuple
- delay/decay/speed/depth gives the delay in milliseconds and
+ Add a chorus to a sound sample. Each quadtuple
+ delay/decay/speed/depth gives the delay in milliseconds and
the decay (relative to gain-in) with a modulation speed in Hz
- using depth in milliseconds. The modulation is either sinu-
- soidal (-s) or triangular (-t). Gain-out is the volume of
+ using depth in milliseconds. The modulation is either sinu-
+ soidal (-s) or triangular (-t). Gain-out is the volume of
the output.
compand attack1,decay1[,attack2,decay2...]
@@ -564,63 +583,63 @@
in-dB1,out-dB1[,in-dB2,out-dB2...]
[gain [initial-volume [delay ] ] ]
- Compand (compress or expand) the dynamic range of a sample.
- The attack and decay time specify the integration time over
+ Compand (compress or expand) the dynamic range of a sample.
+ The attack and decay time specify the integration time over
which the absolute value of the input signal is integrated to
- determine its volume; attacks refer to increases in volume
- and decays refer to decreases. Where more than one pair of
- attack/decay parameters are specified, each channel is
- treated separately and the number of pairs must agree with
+ determine its volume; attacks refer to increases in volume
+ and decays refer to decreases. Where more than one pair of
+ attack/decay parameters are specified, each channel is
+ treated separately and the number of pairs must agree with
the number of input channels. The second parameter is a list
- of points on the compander’s transfer function specified in
- dB relative to the maximum possible signal amplitude. The
- input values must be in a strictly increasing order but the
- transfer function does not have to be monotonically rising.
+ of points on the compander’s transfer function specified in
+ dB relative to the maximum possible signal amplitude. The
+ input values must be in a strictly increasing order but the
+ transfer function does not have to be monotonically rising.
The special value -inf may be used to indicate that the input
volume should be associated output volume. The points
- -inf,-inf and 0,0 are assumed; the latter may be overridden,
+ -inf,-inf and 0,0 are assumed; the latter may be overridden,
but the former may not.
- The third (optional) parameter is a post-processing gain in
- dB which is applied after the compression has taken place;
- the fourth (optional) parameter is an initial volume to be
- assumed for each channel when the effect starts. This per-
- mits the user to supply a nominal level initially, so that,
+ The third (optional) parameter is a post-processing gain in
+ dB which is applied after the compression has taken place;
+ the fourth (optional) parameter is an initial volume to be
+ assumed for each channel when the effect starts. This per-
+ mits the user to supply a nominal level initially, so that,
for example, a very large gain is not applied to initial sig-
nal levels before the companding action has begun to operate:
- it is quite probable that in such an event, the output would
+ it is quite probable that in such an event, the output would
be severely clipped while the compander gain properly adjusts
itself.
- The fifth (optional) parameter is a delay in seconds. The
- input signal is analyzed immediately to control the compan-
- der, but it is delayed before being fed to the volume
- adjuster. Specifying a delay approximately equal to the
- attack/decay times allows the compander to effectively oper-
+ The fifth (optional) parameter is a delay in seconds. The
+ input signal is analyzed immediately to control the compan-
+ der, but it is delayed before being fed to the volume
+ adjuster. Specifying a delay approximately equal to the
+ attack/decay times allows the compander to effectively oper-
ate in a "predictive" rather than a reactive mode.
- copy Copy the input file to the output file. This is the default
+ copy Copy the input file to the output file. This is the default
effect if both files have the same sampling rate.
dcshift shift [ limitergain ]
DC Shift the audio data, with basic linear amplitude formula.
- This is most useful if your audio data tends to not be
- centered around a value of 0. Shifting it back will allow
- you to get the most volume adjustments without clipping audio
+ This is most useful if your audio data tends to not be cen-
+ tered around a value of 0. Shifting it back will allow you
+ to get the most volume adjustments without clipping audio
data.
- The first option is the dcshift value. It is a floating
+ The first option is the dcshift value. It is a floating
point number that indicates the amount to shift.
- An option limtergain value can be specified as well. It
- should have a value much less then 1.0 and is used only on
+ An option limtergain value can be specified as well. It
+ should have a value much less then 1.0 and is used only on
peaks to prevent clipping.
- deemph Apply a treble attenuation shelving filter to samples in
- audio cd format. The frequency response of pre-emphasized
- recordings is rectified. The filtering is defined in the
+ deemph Apply a treble attenuation shelving filter to samples in
+ audio cd format. The frequency response of pre-emphasized
+ recordings is rectified. The filtering is defined in the
standard document ISO 908.
- earwax Makes sound easier to listen to on headphones. Adds audio-
- cues to samples in audio cd format so that when listened to
+ earwax Makes sound easier to listen to on headphones. Adds audio-
+ cues to samples in audio cd format so that when listened to
on headphones the stereo image is moved from inside your head
(standard for headphones) to outside and in front of the lis-
tener (standard for speakers). See
@@ -627,13 +646,13 @@
www.geocities.com/beinges for a full explanation.
echo gain-in gain-out delay decay [ delay decay ... ]
- Add echoing to a sound sample. Each delay/decay part gives
+ Add echoing to a sound sample. Each delay/decay part gives
the delay in milliseconds and the decay (relative to gain-in)
of that echo. Gain-out is the volume of the output.
echos gain-in gain-out delay decay [ delay decay ... ]
- Add a sequence of echos to a sound sample. Each delay/decay
- part gives the delay in milliseconds and the decay (relative
+ Add a sequence of echos to a sound sample. Each delay/decay
+ part gives the delay in milliseconds and the decay (relative
to gain-in) of that echo. Gain-out is the volume of the out-
put.
@@ -647,50 +666,50 @@
volume of the audio from 0 to full volume over fade-in-length
seconds. Specify 0 seconds if no fade-in is wanted.
- For fade-outs, the audio data will be truncated at the stop-
+ For fade-outs, the audio data will be truncated at the stop-
time and the volume will be ramped from full volume down to 0
starting at fade-out-length seconds before the stop-time. No
fade-out is performed if these options are not specified.
- All times can be specified in either periods of time or sam-
- ple counts. To specify time periods use the format
- hh:mm:ss.frac format. To specify using sample counts, spec-
- ify the number of samples and append the letter ’s’ to the
+ All times can be specified in either periods of time or sam-
+ ple counts. To specify time periods use the format
+ hh:mm:ss.frac format. To specify using sample counts, spec-
+ ify the number of samples and append the letter ’s’ to the
sample count (for example 8000s).
An optional type can be specified to change the type of enve-
- lope. Choices are q for quarter of a sinewave, h for half a
- sinewave, t for linear slope, l for logarithmic, and p for
+ lope. Choices are q for quarter of a sinewave, h for half a
+ sinewave, t for linear slope, l for logarithmic, and p for
inverted parabola. The default is a linear slope.
filter [ low ]-[ high ] [ window-len [ beta ] ]
- Apply a Sinc-windowed lowpass, highpass, or bandpass filter
+ Apply a Sinc-windowed lowpass, highpass, or bandpass filter
of given window length to the signal. low refers to the fre-
quency of the lower 6dB corner of the filter. high refers to
the frequency of the upper 6dB corner of the filter.
- A lowpass filter is obtained by leaving low unspecified, or
- 0. A highpass filter is obtained by leaving high unspeci-
- fied, or 0, or greater than or equal to the Nyquist fre-
+ A lowpass filter is obtained by leaving low unspecified, or
+ 0. A highpass filter is obtained by leaving high unspeci-
+ fied, or 0, or greater than or equal to the Nyquist fre-
quency.
The window-len, if unspecified, defaults to 128. Longer win-
- dows give a sharper cutoff, smaller windows a more gradual
+ dows give a sharper cutoff, smaller windows a more gradual
cutoff.
- The beta, if unspecified, defaults to 16. This selects a
+ The beta, if unspecified, defaults to 16. This selects a
Kaiser window. You can select a Nuttall window by specifying
- anything <= 2.0 here. For more discussion of beta, look
+ anything <= 2.0 here. For more discussion of beta, look
under the resample effect.
flanger gain-in gain-out delay decay speed < -s | -t >
Add a flanger to a sound sample. Each triple
- delay/decay/speed gives the delay in milliseconds and the
- decay (relative to gain-in) with a modulation speed in Hz.
- The modulation is either sinodial (-s) or triangular (-t).
+ delay/decay/speed gives the delay in milliseconds and the
+ decay (relative to gain-in) with a modulation speed in Hz.
+ The modulation is either sinodial (-s) or triangular (-t).
Gain-out is the volume of the output.
highp frequency
- Apply a single pole recursive high-pass filter. The fre-
+ Apply a single pole recursive high-pass filter. The fre-
quency response drops logarithmically with I frequency in the
middle of the drop. The slope of the filter is quite gentle.
See filter for a highpass effect with sharper cutoff.
@@ -700,15 +719,12 @@
lowp frequency
Apply a single pole recursive low-pass filter. The frequency
- response drops logarithmically with frequency in the middle
- of the drop. The slope of the filter is quite gentle. See
+ response drops logarithmically with frequency in the middle
+ of the drop. The slope of the filter is quite gentle. See
filter for a lowpass effect with sharper cutoff.
lowpass frequency
Butterworth lowpass filter. Description coming soon!
-
- map Display a list of loops in a sample, and miscellaneous loop
- info.
mask Add "masking noise" to signal. This effect deliberately adds
white noise to a sound in order to mask quantization effects,
--- a/src/sox.c
+++ b/src/sox.c
@@ -94,7 +94,7 @@
static int flow_effect(int);
static int drain_effect(int);
-#define MAX_INPUT_FILES 2
+#define MAX_INPUT_FILES 32
#define MAX_FILES MAX_INPUT_FILES + 1
#ifdef SOXMIX
#define REQUIRED_INPUT_FILES 2
@@ -681,6 +681,14 @@
}
efftab[0].olen = 0;
}
+
+ /* Adjust input side volume based on value specified
+ * by user for this file.
+ */
+ if (file_opts[current_input]->volume != 1.0)
+ clipped += volumechange(efftab[0].obuf,
+ efftab[0].olen,
+ file_opts[current_input]->volume);
#else
for (f = 0; f < input_count; f++)
{@@ -693,11 +701,16 @@
*/
if (ilen[f] == ST_EOF)
ilen[f] = 0;
+
+ /* Adjust input side volume based on value specified
+ * by user for this file.
+ */
+ if (file_opts[f]->volume != 1.0)
+ clipped += volumechange(ibuf[f],
+ ilen[f],
+ file_opts[f]->volume);
}
- /* FIXME: Run threw input data and change its volume
- * based on value in in_file_opts[x]->volume
- */
/* FIXME: Should report if the size of the reads are not
* the same.
*/
@@ -708,18 +721,36 @@
for (s = 0; s < efftab[0].olen; s++)
{- /* Mix data together by dividing by the number
- * of audio files and then summing up. This prevents
- * overflows.
+ /* Mix data together by summing samples together.
+ * It is assumed that input side volume adjustments
+ * will take care of any possible overflow.
+ * By default, SoX sets the volume adjustment
+ * to 1/input_count but the user can override this.
+ * They probably will and some clipping will probably
+ * occur because of this.
*/
for (f = 0; f < input_count; f++)
{if (f == 0)
efftab[0].obuf[s] =
- (s<(st_size_t)ilen[f]) ? (ibuf[f][s]/input_count) : 0;
+ (s<(st_size_t)ilen[f]) ? ibuf[f][s] : 0;
else
if (s < (st_size_t)ilen[f])
- efftab[0].obuf[s] += ibuf[f][s]/input_count;
+ {+ double sample;
+ sample = efftab[0].obuf[s] + ibuf[f][s];
+ if (sample < ST_SAMPLE_MIN)
+ {+ sample = ST_SAMPLE_MIN;
+ clipped++;
+ }
+ else if (sample > ST_SAMPLE_MAX)
+ {+ sample = ST_SAMPLE_MAX;
+ clipped++;
+ }
+ efftab[0].obuf[s] = sample;
+ }
}
}
#endif
@@ -1196,12 +1227,12 @@
top = buf+ct;
while (p < top) {y = vol * *p;
- if (y < -2147483647.0) {- y = -2147483647.0;
+ if (y < ST_SAMPLE_MIN) {+ y = ST_SAMPLE_MIN;
clips++;
}
- else if (y > 2147483647.0) {- y = 2147483647.0;
+ else if (y > ST_SAMPLE_MAX) {+ y = ST_SAMPLE_MAX;
clips++;
}
*p++ = y + 0.5;