ref: 7e27d2ef6a96c866715f1e9ccf505b89991e1bee
parent: f5fbe6ce3c8f779c4432203968ec4cc808588859
author: robs <robs>
date: Sun Aug 31 15:51:34 EDT 2008
stat doc updates
--- a/ChangeLog
+++ b/ChangeLog
@@ -44,6 +44,7 @@
o Fix memory leaks in effects chain when restarting effects.
o Fix WAV write on 64-bit arch. (robs)
o Fixed pkg-config CFLAGS. (evilynux)
+ o Improved documentation for the `stat' effect. (robs)
Internal improvements:
--- a/soxeffect.7
+++ b/soxeffect.7
@@ -1483,34 +1483,72 @@
.I leeway
should be set to zero.
.TP
-\fBstat\fR [\fB\-s \fIn\fR] [\fB\-rms\fR] [\fB\-freq\fR] [\fB\-v\fR] [\fB\-d\fR]
-Do a statistical check on the input file,
-and print results on the standard error file. Audio is passed
-unmodified through the SoX processing chain.
+\fBstat\fR [\fB\-s \fIscale\fR] [\fB\-rms\fR] [\fB\-freq\fR] [\fB\-v\fR] [\fB\-d\fR]
+Display time and frequency domain statistical information about the audio.
+Audio is passed unmodified through the SoX processing chain.
.SP
-The `Volume Adjustment:' field in the statistics
-gives you the parameter to the
-.B \-v
-.I number
-which will make the audio as loud as possible without clipping.
+The information is output to the `standard error' (stderr) stream and is
+calculated, where
+.I n
+is the duration of the audio in samples,
+.I c
+is the number of audio channels,
+.I r
+is the audio sample rate, and
+.I x\s-2\dk\u\s0
+represents the PCM value (in the range \-1 to +1 by default) of each successive
+sample in the audio,
+as follows:
+.TS
+center;
+lI l l.
+Samples read \fIn\fR\^\(mu\^\fIc\fR \
+Length (seconds) \fIn\fR\^\(di\^\fIr\fR
+Scaled by \ See \-s below.
+Maximum amplitude max(\fIx\s-2\dk\u\s0\fR) T{+The maximum sample value in the audio; usually this will be a positive number.
+T}
+Minimum amplitude min(\fIx\s-2\dk\u\s0\fR) T{+The minimum sample value in the audio; usually this will be a negative number.
+T}
+Midline amplitude \(12\^min(\fIx\s-2\dk\u\s0\fR)\^+\^\(12\^max(\fIx\s-2\dk\u\s0\fR)
+Mean norm \(S1/\s-2n\s+2\^\(*S\^\^\(br\^\fIx\s-2\dk\u\s0\fR\^\(br\^ T{+The average of the absolute value of each sample in the audio.
+T}
+Mean amplitude \(S1/\s-2n\s+2\^\(*S\^\fIx\s-2\dk\u\s0\fR T{+The average of each sample in the audio. If this figure is non-zero, then it indicates the
+presence of a D.C. offset (which could be removed using the
+.B dcshift
+effect).
+T}
+RMS amplitude \(sr(\(S1/\s-2n\s+2\^\(*S\^\fIx\s-2\dk\u\s0\fR\(S2) T{+The level of a D.C. signal that would have the same power
+as the audio's average power.
+T}
+Maximum delta max(\^\(br\^\fIx\s-2\dk\u\s0\fR\^\-\^\fIx\s-2\dk\-1\u\s0\fR\^\(br\^)
+Minimum delta min(\^\(br\^\fIx\s-2\dk\u\s0\fR\^\-\^\fIx\s-2\dk\-1\u\s0\fR\^\(br\^)
+Mean delta \(S1/\s-2n\-1\s+2\^\(*S\^\^\(br\^\fIx\s-2\dk\u\s0\fR\^\-\^\fIx\s-2\dk\-1\u\s0\fR\^\(br\^
+RMS delta \(sr(\(S1/\s-2n\-1\s+2\^\(*S\^(\fIx\s-2\dk\u\s0\fR\^\-\^\fIx\s-2\dk\-1\u\s0\fR)\(S2)
+Rough frequency \ In Hz.
+Volume Adjustment \ T{+The parameter to the
+.B vol
+effect which would make the audio as loud as possible without clipping.
Note: See the discussion on
.B Clipping
in
.BR sox (1)
-for reasons why it is rarely a good idea to actually do this.
+for reasons why it is rarely a good idea actually to do this.
+T}
+.TE
+.DT
.SP
-The option
-.B \-v
-will print out the `Volume Adjustment:' field's value only and
-return. This could be of use in scripts to auto convert the
-volume.
-.SP
The
.B \-s
-option is used to scale the input data by a given factor. The default value
-of
-.I n
-is the maximum value of a signed long integer (7fffffff in hexadecimal).
+option can be used to scale the input data by a given factor.
+The default value of
+.I scale
+is 2147483647 (i.e. the maximum value of a 32-bit signed integer).
Internal effects
always work with signed long PCM data and so the value should relate to this
fact.
@@ -1521,15 +1559,21 @@
format.
.SP
The
+.B \-v
+option displays only the `Volume Adjustment' value.
+.SP
+The
.B \-freq
-option calculates the input's power spectrum and prints it to standard error.
+option calculates the input's power spectrum (4096 point DFT) instead of the
+statistics listed above.
.SP
-There is also an optional parameter
+The
.B \-d
-that will print out a hex dump of the
-audio from the internal buffer that is in 32-bit signed PCM data.
-This is mainly only of use in tracking down endian problems that
-creep in to SoX on cross-platform versions.
+option
+displays a hex dump of the 32-bit signed PCM data
+audio in SoX's internal buffer.
+This is mainly used to help track down endian problems that
+sometimes occur in cross-platform versions of SoX.
.TP
\fBswap\fR [\fI1 2\fR | \fI1 2 3 4\fR]
Swap channels in multi-channel audio files. Optionally, you may
--
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