[PATCH]: typos etc. in manual page lame.1
Urs Thuermann <[email protected]>
| Newsgroups | gmane.comp.audio.mp3.lame |
|---|---|
| Message-ID | <[email protected]> |
This patch fixes a number of typos, wordings, font selection for puntuators etc. I'd like it to be applied to the lame sources. I'm not on the mailing list. Please CC any answers to me. urs diff -ru lame-3.99.5.orig/doc/man/lame.1 lame-3.99.5/doc/man/lame.1 --- lame-3.99.5.orig/doc/man/lame.1 2011-10-02 16:52:19.000000000 +0200 +++ lame-3.99.5/doc/man/lame.1 2012-04-11 20:16:09.000000000 +0200 @@ -2,15 +2,19 @@ .SH NAME lame \- create mp3 audio files .SH SYNOPSIS -lame [options] <infile> <outfile> +.B lame +.RI [ options ] +.I infile outfile .SH DESCRIPTION .PP LAME is a program which can be used to create compressed audio files. (Lame ain't an MP3 encoder). These audio files can be played back by popular MP3 players such as mpg123 or madplay. -To read from stdin, use "\-" for <infile>. -To write to stdout, use "\-" for <outfile>. +To read from stdin, use "\-" for +.IR infile . +To write to stdout, use "\-" for +.IR outfile . .SH OPTIONS Input options: .TP @@ -21,7 +25,7 @@ first, then right channel. By default, LAME expects them to be signed integers with a bitwidth of 16. Without -.B \-r, +.BR \-r , LAME will perform several .I fseek()'s on the input file looking for WAV and AIFF headers. @@ -30,14 +34,14 @@ .TP .B \-x Swap bytes in the input file or output file when using -.B \-\-decode. +.BR \-\-decode . .br For sorting out little endian/big endian type problems. If your encodings sounds like static, try this first. .br Without using -.B \-x, +.BR \-x , LAME will treat input file as native endian. .TP .BI \-s " sfreq" @@ -125,7 +129,7 @@ the two input channels. It can, however, -negotiate the bit demand between both channel, +negotiate the bit demand between both channels, i.e. give one channel more bits if the other contains silence or needs less bits because of a lower complexity. @@ -144,7 +148,7 @@ Using mid/side stereo inappropriately can result in audible compression artifacts. -To much switching between mid/side and regular stereo can also +Too much switching between mid/side and regular stereo can also sound bad. To determine when to switch to mid/side stereo, LAME uses a much more sophisticated algorithm than that described @@ -201,7 +205,7 @@ use .B lame \-m .I s -.B \-a. +.BR \-a . For WAV and AIFF input files, using @@ -319,7 +323,7 @@ it is a value stored in the header of a sound file. Information on the purpose of ReplayGain and the algorithms used is available from -.B http://www.replaygain.org/. +.BR http://www.replaygain.org/ . Only the "RadioGain" Replaygain value is computed, it is stored in the LAME tag. @@ -331,7 +335,7 @@ This switch is enabled by default. See also: -.B \-\-replaygain\-accurate, \-\-noreplaygain +.BR \-\-replaygain\-accurate ", " \-\-noreplaygain .TP .B \-\-replaygain\-accurate Compute ReplayGain more accurately and find the peak sample. @@ -346,7 +350,7 @@ it is a value stored in the header of a sound file. Information on the purpose of ReplayGain and the algorithms used is available from -.B http://www.replaygain.org/. +.BR http://www.replaygain.org/ . By default, LAME performs ReplayGain analysis on the input data @@ -399,7 +403,7 @@ volume scaling). See also: -.B \-\-replaygain-fast, \-\-noreplaygain \-\-clipdetect +.BR \-\-replaygain-fast ", " \-\-noreplaygain\ \-\-clipdetect .RE .TP .B \-\-noreplaygain @@ -408,7 +412,7 @@ By default ReplayGain analysis is enabled. This switch disables it. See also: -.B \-\-replaygain-fast, \-\-replaygain-accurate +.BR \-\-replaygain-fast ", " \-\-replaygain-accurate .TP .B \-\-clipdetect Clipping detection. @@ -457,7 +461,7 @@ .B \-\-nohist By default, LAME will display a bitrate histogram while producing VBR mp3 files. -This will disable that feature. +This option will disable that feature. .br Histogram display might not be available on your release. .TP @@ -488,10 +492,10 @@ The higher the bitrate, the higher the quality. But for a given bitrate, -we have a choice of algorithms to determine the best scalefactors +we have a choice of algorithms to determine the best scale factors and Huffman encoding (noise shaping). -.B -q 0: +.BR "-q 0" : .br use slowest & best possible version of all algorithms. .B -q 0 @@ -499,28 +503,28 @@ .B -q 1 are slow and may not produce significantly higher quality. -.B -q 2: +.BR "-q 2" : .br recommended. Same as -.B -h. +.BR -h . -.B -q 5: +.BR "-q 5" : .br default value. Good speed, reasonable quality. -.B -q 7: +.BR "-q 7": .br same as -.B -f. +.BR -f . Very fast, ok quality. Psycho acoustics are used for pre-echo & M/S, but no noise shaping is done. -.B -q 9: +.BR "-q 9" : .br disables almost all algorithms including psy-model. Poor quality. @@ -609,7 +613,7 @@ .TP .B -v use variable bitrate -.B (\-\-vbr-new) +.RB ( \-\-vbr-new ) .TP .B \-\-vbr-old Invokes the oldest, @@ -623,11 +627,11 @@ .B \-\-vbr-new Invokes the newest VBR algorithm. During the development of version 3.90, -considerable tuning was done on this algorithm, +considerable tuning has been done on this algorithm, and it is now considered to be on par with the original -.B \-\-vbr-old. -It has the added advantage of being very fast (over twice as fast as -.B \-\-vbr-old). +.BR \-\-vbr-old . +It has the added advantage of being very fast (more than twice as fast as +.BR \-\-vbr-old ). .TP .BI -V " n" 0 <= @@ -705,12 +709,12 @@ <= 7 When LAME searches for a "good" quantization, -it has to compare the actual one with the best one found so far. +it has to compare the current one with the best one found so far. The comparison says which one is better, -the best so far or the actual. +the best so far or the current. The .B -X -parameter selects between different approaches to make this decision, +parameter selects among different approaches to make this decision, .B -X0 being the default mode: @@ -718,7 +722,7 @@ .br The criteria are (in order of importance): .br -* less distorted scalefactor bands +* less distorted scale factor bands .br * the sum of noise over the thresholds is lower .br @@ -726,18 +730,18 @@ .B -X1 .br -The actual is better if the maximum noise over all scalefactor bands is +The current is better if the maximum noise over all scale factor bands is less than the best so far. .B -X2 .br -The actual is better if the total sum of noise is lower than the best so +The current is better if the total sum of noise is lower than the best so far. .B -X3 .br -The actual is better if the total sum of noise is lower than the best so -far and the maximum noise over all scalefactor bands is less than the +The current is better if the total sum of noise is lower than the best so +far and the maximum noise over all scale factor bands is less than the best so far plus 2dB. .B -X4 @@ -758,7 +762,7 @@ .br * the sum of noise over the thresholds is lower .br -* the maximum noise over all scalefactor bands is lower +* the maximum noise over all scale factor bands is lower .br * the total sum of noise is lower @@ -766,7 +770,7 @@ .br The criteria are: .br -* less distorted scalefactor bands +* less distorted scale factor bands .br or .br @@ -798,7 +802,7 @@ A better solution would be to apply the de-emphasis with a standalone utility before encoding, and then encode without -.B -e. +.BR -e . .TP .B -c Mark the encoded file as being copyrighted. @@ -917,8 +921,10 @@ Analysis options: .TP .B \-g -run graphical analysis on <infile>. -<infile> can also be a .mp3 file. +run graphical analysis on +.IR infile . +.I infile +can also be a .mp3 file. (This feature is a compile time option. Your binary may for speed reasons be compiled without this.) @@ -952,17 +958,17 @@ the encoder will use the same bitrate, so the quality of your mp3 is variable. Complex parts will be of a lower quality than the easiest ones. -The main advantage is that the final files size won't change and +The main advantage is that the final file size won't change and can be accurately predicted. .TP .B Average Bitrate (ABR) In this mode, -you choose the encoder will maintain an average bitrate while using +the encoder will maintain an average bitrate while using higher bitrates for the parts of your music that need more bits. The result will be of higher quality than CBR encoding but the average file size will remain predictable, so this mode is highly recommended over CBR. -This encoding mode is similar to what is referred as vbr in AAC or +This encoding mode is similar to what is referred to as vbr in AAC or Liquid Audio (2 other compression technologies). .TP .B Variable bitrate (VBR) @@ -972,8 +978,8 @@ Then encoder tries to maintain the given quality in the whole file by choosing the optimal number of bits to spend for each part of your music. The main advantage is that you are able to specify the quality level that -you want to reach, -but the inconvenient is that the final file size is totally unpredictable. +you want to achieve, +but the inconvenience is that the final file size is totally unpredictable. .SH PRESETS The @@ -1003,7 +1009,7 @@ .TP .B \-\-preset insane This preset will usually be overkill for most people and most situations, -but if you must have the absolute highest quality with no regard to filesize, +but if you must have the absolutely highest quality with no regard to file size, this is the way to go. .PP For ABR modes (high quality per given bitrate but not as high as VBR): @@ -1120,8 +1126,8 @@ .PP Probably there are some. .SH SEE ALSO -.BR mpg123 (1) , -.BR madplay (1) , +.BR mpg123 (1), +.BR madplay (1), .BR sox (1) .SH AUTHORS .nf ------------------------------------------------------------------------------ Better than sec? 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