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Old 04-02-2006, 04:04 AM   #1
wibbly
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Default LAME settings

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Anyone got LAME settings (command line) known to be appropriate for generating small 8700 compatible mp3's?

... Having them would save (me) a load of experimentation, and as LAME is free (and also works to re-encode existing MP3s, or WAVs) it would seem a easy target for a trivial batch file called 'make_8700_mp3.bat"

W
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Old 04-02-2006, 08:45 AM   #2
rcbjr
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Quote:
Originally Posted by wibbly
Anyone got LAME settings (command line) known to be appropriate for generating small 8700 compatible mp3's?

... Having them would save (me) a load of experimentation, and as LAME is free (and also works to re-encode existing MP3s, or WAVs) it would seem a easy target for a trivial batch file called 'make_8700_mp3.bat"

---

What parameters are you looking for? I use Audacity, which requires the Lame Dll for exporting MP3s. It only requires one additional parameter, the BIT RATE. I use 56 (K I assume).

Hope this helps.
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Old 04-02-2006, 09:16 AM   #3
taylor
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Or you could try DBPower Amp.
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Old 04-02-2006, 01:29 PM   #4
wibbly
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Well, since you ask... Most of these have got to be irrelevant to the task at hand, but what's best to use for [options] for generating 8700 compatible mp3's?

===

LAME version 3.90.3 MMX (http://www.mp3dev.org/)
-- Compiled at http://www.hydrogenaudio.org
-- Check this website for up to date information on the --alt-presets

usage: C:\Program Files\Lame\lame.exe [options] <infile> [outfile]

<infile> and/or <outfile> can be "-", which means stdin/stdout.

RECOMMENDED:
lame -h input.wav output.mp3

OPTIONS:
Input options:
-r input is raw pcm
-x force byte-swapping of input
-s sfreq sampling frequency of input file (kHz) - default 44.1 kHz
--bitwidth w input bit width is w (default 16)
--mp1input input file is a MPEG Layer I file
--mp2input input file is a MPEG Layer II file
--mp3input input file is a MPEG Layer III file
--nogap <file1> <file2> <...>
gapless encoding for a set of contiguous files
--nogapout <dir>
output dir for gapless encoding (must precede --nogap)

Operational options:
-m <mode> (s)tereo, (j)oint, (f)orce, (m)ono or (a)auto
default is (s) or (j) depending on bitrate
force = force ms_stereo on all frames.
auto = jstereo, with varialbe mid/side threshold
-a downmix from stereo to mono file for mono encoding
-d allow channels to have different blocktypes
--freeformat produce a free format bitstream
--decode input=mp3 file, output=wav
-t disable writing wav header when using --decode
--comp <arg> choose bitrate to achive a compression ratio of <arg>
--scale <arg> scale input (multiply PCM data) by <arg>
--scale-l <arg> scale channel 0 (left) input (multiply PCM data) by <arg>
--scale-r <arg> scale channel 1 (right) input (multiply PCM data) by <arg>
--preset type type must be phone, voice, fm, tape, hifi, cd or studio
"--preset help" gives some more infos on these
--alt-preset type type must be "standard", "extreme", "insane",
or a value for an average desired bitrate and depending on

the value specified, appropriate quality settings will be
used.
--r3mix use high-quality VBR preset

Verbosity:
--disptime <arg>print progress report every arg seconds
-S don't print progress report, VBR histograms
--nohist disable VBR histogram display
--silent don't print anything on screen
--quiet don't print anything on screen
--verbose print a lot of useful information

Noise shaping & psycho acoustic algorithms:
-q <arg> <arg> = 0...9. Default -q 5
-q 0: Highest quality, very slow
-q 9: Poor quality, but fast
-h Same as -q 2. Recommended.
-f Same as -q 7. Fast, ok quality


CBR (constant bitrate, the default) options:
-b <bitrate> set the bitrate in kbps, default 128 kbps

ABR options:
--abr <bitrate> specify average bitrate desired (instead of quality)

VBR options:
-v use variable bitrate (VBR) (--vbr-old)
--vbr-old use old variable bitrate (VBR) routine
--vbr-new use new variable bitrate (VBR) routine
-V n quality setting for VBR. default n=4
0=high quality,bigger files. 9=smaller files
-b <bitrate> specify minimum allowed bitrate, default 32 kbps
-B <bitrate> specify maximum allowed bitrate, default 320 kbps
-F strictly enforce the -b option, for use with players that
do not support low bitrate mp3
-t disable writing LAME Tag


ATH related:
--noath turns ATH down to a flat noise floor
--athshort ignore GPSYCHO for short blocks, use ATH only
--athonly ignore GPSYCHO completely, use ATH only
--athtype n selects between different ATH types [0-5]
--athlower x lowers ATH by x dB
--athaa-type n ATH auto adjust types 1-3, else no adjustment
--athaa-loudapprox n n=1 total energy or n=2 equal loudness curve
--athaa-sensitivity x activation offset in -/+ dB for ATH auto-adjustment

PSY related:
--short use short blocks when appropriate
--noshort do not use short blocks
--allshort use only short blocks
--cwlimit <freq> compute tonality up to freq (in kHz) default 8.8717
--notemp disable temporal masking effect
--nspsytune experimental PSY tunings by Naoki Shibata
--nssafejoint M/S switching criterion
--nsmsfix <arg> M/S switching tuning [effective 0-3.5]
--ns-bass x adjust masking for sfbs 0 - 6 (long) 0 - 5 (short)
--ns-alto x adjust masking for sfbs 7 - 13 (long) 6 - 10 (short)
--ns-treble x adjust masking for sfbs 14 - 21 (long) 11 - 12 (short)
--ns-sfb21 x change ns-treble by x dB for sfb21


experimental switches:
-X n selects between different noise measurements
-Y lets LAME ignore noise in sfb21, like in CBR


MP3 header/stream options:
-e <emp> de-emphasis n/5/c (obsolete)
-c mark as copyright
-o mark as non-original
-p error protection. adds 16 bit checksum to every frame
(the checksum is computed correctly)
--nores disable the bit reservoir
--strictly-enforce-ISO comply as much as possible to ISO MPEG spec

Filter options:
-k keep ALL frequencies (disables all filters),
Can cause ringing and twinkling
--lowpass <freq> frequency(kHz), lowpass filter cutoff above freq
--lowpass-width <freq> frequency(kHz) - default 15% of lowpass freq
--highpass <freq> frequency(kHz), highpass filter cutoff below freq
--highpass-width <freq> frequency(kHz) - default 15% of highpass freq
--resample <sfreq> sampling frequency of output file(kHz)- default=automatic


ID3 tag options:
--tt <title> audio/song title (max 30 chars for version 1 tag)
--ta <artist> audio/song artist (max 30 chars for version 1 tag)
--tl <album> audio/song album (max 30 chars for version 1 tag)
--ty <year> audio/song year of issue (1 to 9999)
--tc <comment> user-defined text (max 30 chars for v1 tag, 28 for v1.1)
--tn <track> audio/song track number (1 to 255, creates v1.1 tag)
--tg <genre> audio/song genre (name or number in list)
--add-id3v2 force addition of version 2 tag
--id3v1-only add only a version 1 tag
--id3v2-only add only a version 2 tag
--space-id3v1 pad version 1 tag with spaces instead of nulls
--pad-id3v2 pad version 2 tag with extra 128 bytes
--genre-list print alphabetically sorted ID3 genre list and exit

Note: A version 2 tag will NOT be added unless one of the input fields
won't fit in a version 1 tag (e.g. the title string is longer than 30
characters), or the '--add-id3v2' or '--id3v2-only' options are used,
or output is redirected to stdout.


MPEG-1 layer III sample frequencies (kHz): 32 48 44.1
bitrates (kbps): 32 40 48 56 64 80 96 112 128 160 192 224 256 320

MPEG-2 layer III sample frequencies (kHz): 16 24 22.05
bitrates (kbps): 8 16 24 32 40 48 56 64 80 96 112 128 144 160

MPEG-2.5 layer III sample frequencies (kHz): 8 12 11.025
bitrates (kbps): 8 16 24 32 40 48 56 64 80 96 112 128 144 160
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Old 04-02-2006, 03:49 PM   #5
rcbjr
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The only onee that I changed ( to reduce the file size) in audacity is the MP3 export bit rate, which would be -b. The Defaults listed for the inputs should work, although I have the default Sample Format set to 32-Bit Float. Have you tried it yet? Does it work? If so don't mess with the settings.

Hope this helps.
Rcbjr
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