Audiograbber Help Please!

[zen]

I've just discovered I hav to rerip my cd collection because of space issues. I want to make them vbr (128 to 192), joint stereo. I can't work out how to get these settings on Audiograbber, it has a load of settings like vbr 0, 1, 2 etc. and other stuff I have no idea about...

any help would be great...I need to rip them right this time!

[Dunno]

Here's a description for all the various switches of LAME mp3,

Full command line switch reference
note: Options which could exist without beeing documented here are considered as experimental ones. Such experimental options should usually not be used.
switch parameter
-a
downmix stereo file to mono
--abr
average bitrate encoding
--allshort
use short blocks only
--athlower
lower the ATH
--athonly
ATH only
--athshort
ATH only for short blocks
--athtype
select ATH type
-b
bitrate (8...320)
-B
max VBR/ABR bitrate (8...320)
--bitwidth
input bit width
-c
copyright
--cbr
enforce use of constant bitrate
--comp
choose compression ratio
--cwlimit
tonality limit
-d
block type control
--decode
decoding only
--disptime
time between display updates
-e
de-emphasis (n, 5, c)
-f
fast mode
-F
strictly enforce the -b option
--freeformat
free format bitstream
-h
high quality
--help
help
--highpass
highpass filtering frequency in kHz
--highpass-width
width of highpass filtering in kHz
-k
full bandwidth
--lowpass
lowpass filtering frequency in kHz
--lowpass-width
width of lowpass filtering in kHz
-m
stereo mode (s, j, f, m)
--mp1input
MPEG Layer I input file
--mp2input
MPEG Layer II input file
--mp3input
MPEG Layer III input file
--noath
disable ATH
--noasm
disable assembly optimizations (mmx/3dnow/sse)
--nohist
disable histogram display
--nores
disable bit reservoir
--noshort
disable short blocks frames
--notemp
disable temporal masking
-o
non-original
-p
error protection
--preset
use built-in preset
--alt-preset
use updated and much higher quality "alternate" presets
--priority
OS/2 process priority control
-q
algorithm quality selection
--quiet
silent operation
-r
input file is raw pcm
--resample
output sampling frequency in kHz (encoding only)
--r3mix
r3mix VBR preset
-s
sampling frequency in kHz
-S
silent operation
--scale
scale input
--scale-l
scale input channel 0 (left)
--scale-r
scale input channel 1 (right)
--short
use short blocks
--silent
silent operation
--strictly-enforce-ISO
strict ISO compliance
-t
disable INFO/WAV header
-V
VBR quality setting (0...9)
--vbr-new
new VBR mode
--vbr-old
older VBR mode
--verbose
verbosity
-x
swapbytes
-X
change quality measure

* -a downmix
Mix the stereo input file to mono and encode as mono.
The downmix is calculated as the sum of the left and right channel, attenuated by 6 dB.

This option is only needed in the case of raw PCM stereo input (because LAME cannot determine the number of channels in the input file).
To encode a stereo PCM input file as mono, use "lame -m s -a".

For WAV and AIFF input files, using "-m m" will always produce a mono .mp3 file from both mono and stereo input.



* --abr n average bitrate encoding
Turns on encoding with a targeted average bitrate of n kbits, allowing to use frames of different sizes. The allowed range of n is 8-310, you can use any integer value within that range.

It can be combined with the -b and -B switches like:
lame --abr 123 -b 64 -B 192 a.wav a.mp3
which would limit the allowed frame sizes between 64 and 192 kbits.


* --allshort use short blocks only
Use only short blocks, no long ones.




* --athlower n lower the ATH
Lower the ATH (absolute threshold of hearing) by n dB.
Normally, humans are unable to hear any sound below this threshold, but for music recorded at very low level this option might be usefull.




* --athonly ATH only
This option causes LAME to ignore the output of the psy-model and only use masking from the ATH (absolute threshold of hearing). Might be useful at very high bitrates or for testing the ATH.




* --athshort ATH only for short blocks
Ignore psychoacoustic model for short blocks, use ATH only.




* --athtype 0/1/2 select ATH type
The Absolute Threshold of Hearing is the minimum threshold under which humans are unable to hear any sound. In the past, LAME was using ATH shape 0 which is the Painter & Spanias formula. Tests have shown that this formula is innacurate for the 13-22 kHz area, leading to audible artifacts in some cases. Shape 1 was thus implemented, which is over sensitive, leading to very high bitrates. Shape 2 formula was accurately modelized from real data in order to real optimal quality while not wasting bitrate. In CBR and ABR modes, LAME uses ATH shape 2 by default.

In VBR mode, LAME is adapting its shape according to the -V value, going gradually from the 0 shape at -V9 up to shape 2 at -V0.




* -b n bitrate
For MPEG1 (sampling frequencies of 32, 44.1 and 48 kHz)
n = 32,40,48,56,64,80,96 ,112,128,160,192,224 ,256,320

For MPEG2 (sampling frequencies of 16, 22.05 and 24 kHz)
n = 8,16,24,32,40,48,56, 64,80,96,112,128,144 ,160

Default is 128 kbs for MPEG1 and 64 kbs for MPEG2.

When used with variable bitrate encoding (VBR), -b specifies the minimum bitrate to be used. However, in order to avoid wasted space, the smallest frame size available will be used during silences.



* -B n maximum VBR/ABR bitrate
For MPEG1 (sampling frequencies of 32, 44.1 and 48 kHz)
n = 32,40,48,56,64,80,96 ,112,128,160,192,224 ,256,320

For MPEG2 (sampling frequencies of 16, 22.05 and 24 kHz)
n = 8,16,24,32,40,48,56, 64,80,96,112,128,144 ,160

Specifies the maximum allowed bitrate when using VBR/ABR

The use of -B is NOT RECOMMENDED. A 128kbs CBR bitstream, because of the bit reservoir, can actually have frames which use as many bits as a 320kbs frame. VBR modes minimize the use of the bit reservoir, and thus need to allow 320kbs frames to get the same flexibility as CBR streams.

note: If you own an mp3 hardware player build upon a MAS 3503 chip, you must set maximum bitrate to no more than 224 kpbs.
* --bitwidth 8/16/24/32 input bit width
Required only for raw PCM input files. Otherwise it will be determined from the header of the input file.


* -c copyright
Mark the encoded file as being copyrighted.



* --cbr enforce use of constant bitrate
This switch enforces the use of constant bitrate encoding.



* --comp choose compression ratio
Instead of choosing bitrate, using this option, user can choose compression ratio to achieve.



* --cwlimit n tonality limit
Compute tonality up to freq (in kHz). Default setting is 8.8717.



* -d block type control
Allows the left and right channels to use different block size types.



* --decode decoding only
Uses LAME for decoding to a wav file. The input file can be any input type supported by encoding, including layer I,II,III (MP3) and OGG files. In case of MPEG files, LAME uses a bugfixed version of mpglib for decoding.

If -t is used (disable wav header), Lame will output raw pcm in native endian format. You can use -x to swap bytes order.



* --disptime n time between display updates
Set the delay in seconds between two display updates.



* -e n/5/c de-emphasis

n = (none, default)
5 = 0/15 microseconds
c = citt j.17

All this does is set a flag in the bitstream. If you have a PCM input file where one of the above types of (obsolete) emphasis has been applied, you can set this flag in LAME. Then the mp3 decoder should de-emphasize the output during playback, although most decoders ignore this flag.

A better solution would be to apply the de-emphasis with a standalone utility before encoding, and then encode without -e.



* -f fast mode
This switch forces the encoder to use a faster encoding mode, but with a lower quality. The behaviour is the same as the -q7 switch.

Noise shaping will be disabled, but psycho acoustics will still be computed for bit allocation and pre-echo detection.



* -F strictly enforce the -b option
This is mainly for use with hardware players that do not support low bitrate mp3.

Without this option, the minimum bitrate will be ignored for passages of analog silence, ie when the music level is below the absolute threshold of human hearing (ATH).



* --freeformat free format bitstream
Produces a free format bitstream. With this option, you can use -b with any bitrate higher than 8 kbps.

However, even if an mp3 decoder is required to support free bitrates at least up to 320 kbps, many players are unable to deal with it.

Tests have shown that the following decoders support free format:

FreeAmp up to 440 kbps
in_mpg123 up to 560 kbps
l3dec up to 310 kbps
LAME up to 560 kbps
MAD up to 640 kbps



* -h high quality
Use some quality improvements. Encoding will be slower, but the result will be of higher quality. The behaviour is the same as the -q2 switch.
This switch is always enabled when using VBR.



* --help help
Display a list of all available options.



* --highpass highpass filtering frequency in kHz
Set an highpass filtering frequency. Frequencies below the specified one will be cutoff.



* --highpass-width width of highpass filtering in kHz
Set the width of the highpass filter. The default value is 15% of the highpass frequency.



* -k full bandwidth
Tells the encoder to use full bandwidth and to disable all filters. By default, the encoder uses some highpass filtering at low bitrates, in order to keep a good quality by giving more bits to more important frequencies.
Increasing the bandwidth from the default setting might produce ringing artefacts at low bitrates. Use with care!



* --lowpass lowpass filtering frequency in kHz
Set a lowpass filtering frequency. Frequencies above the specified one will be cutoff.



* --lowpass-width width of lowpass filtering in kHz
Set the width of the lowpass filter. The default value is 15% of the lowpass frequency.



* -m s/j/f/d/m stereo mode
Joint-stereo is the default mode for stereo files with VBR when -V is more than 4 or fixed bitrates of 160kbs or less. At higher fixed bitrates or higher VBR settings, the default is stereo.

stereo
In this mode, the encoder makes no use of potentially existing correlations between the two input channels. It can, however, negotiate the bit demand between both channel, i.e. give one channel more bits if the other contains silence or needs less bits because of a lower complexity.

joint stereo
In this mode, the encoder will make use of a correlation between both channels. The signal will be matrixed into a sum ("mid"), computed by L+R, and difference ("side") signal, computed by L-R, and more bits are allocated to the mid channel.
This will effectively increase the bandwidth if the signal does not have too much stereo separation, thus giving a significant gain in encoding quality.

Using mid/side stereo inappropriately can result in audible compression artifacts. To 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 in the ISO documentation, and thus is safe to use in joint stereo mode.

forced joint stereo
This mode will force MS joint stereo on all frames. It's slightly faster than joint stereo, but it should be used only if you are sure that every frame of the input file has very little stereo separation.

dual channels
In this mode, the 2 channels will be totally indenpendently encoded. Each channel will have exactly half of the bitrate. This mode is designed for applications like dual languages encoding (ex: English in one channel and French in the other). Using this encoding mode for regular stereo files will result in a lower quality encoding.

mono
The input will be encoded as a mono signal. If it was a stereo signal, it will be downsampled to mono. The downmix is calculated as the sum of the left and right channel, attenuated by 6 dB.



* --mp1input MPEG Layer I input file
Assume the input file is a MPEG Layer I file.
If the filename ends in ".mp1" or ".mpg" LAME will assume it is a MPEG Layer I file. For stdin or Layer I files which do not end in .mp1 or .mpg you need to use this switch.



* --mp2input MPEG Layer II input file
Assume the input file is a MPEG Layer II (ie MP2) file.
If the filename ends in ".mp2" LAME will assume it is a MPEG Layer II file. For stdin or Layer II files which do not end in .mp2 you need to use this switch.



* --mp3input MPEG Layer III input file
Assume the input file is a MP3 file. Usefull for downsampling from one mp3 to another. As an example, it can be usefull for streaming through an IceCast server.
If the filename ends in ".mp3" LAME will assume it is an MP3 file. For stdin or MP3 files which do not end in .mp3 you need to use this switch.



* --noath disable ATH
Disable any use of the ATH (absolute threshold of hearing) for masking. Normally, humans are unable to hear any sound below this threshold.



* --noasm mmx/3dnow/sse disable assembly optimisations
Disable specific assembly optimizations. Quality will not increase, only speed will be reduced. If you have problems running Lame on a Cyrix/Via processor, disabling mmx optimizations might solve your problem.



* --nohist disable histogram display
By default, LAME will display a bitrate histogram while producing VBR mp3 files. This will disable that feature.
Histogram display might not be available on your release.



* --nores disable bit reservoir
Disable the bit reservoir. Each frame will then become independent from previous ones, but the quality will be lower.



* --noshort disable short blocks frames
Encode all frames using long blocks only. This could increase quality when encoding at very low bitrates, but can produce serious pre-echo artefacts.



* --notemp disable temporal masking
Don't make use of the temporal masking effect.



* -o non-original
Mark the encoded file as being a copy.



* -p error protection
Turn on CRC error protection.
It will add a cyclic redundancy check (CRC) code in each frame, allowing to detect transmission errors that could occur on the MP3 stream. However, it takes 16 bits per frame that would otherwise be used for encoding, and then will slightly reduce the sound quality.



* --preset presetName use built-in preset
Use one of the built-in presets (phone, phon+, lw, mw-eu, mw-us, sw, fm, voice, radio, tape, hifi, cd, studio).
"--preset help" gives more information about the used options in these presets.



* --alt-preset presetName use updated and much higher quality "alternate" presets
Use one of the built-in alternate presets (standard, fast standard, extreme, fast extreme, insane, or the abr/cbr modes).
"--alt-preset help" gives more information about the usage possibilities for these presets.



* --priority 0...4 OS/2 process priority control
With this option, LAME will run with a different process priority under IBM OS/2.
This will greatly improve system responsiveness, since OS/2 will have more free time to properly update the screen and poll the keyboard/mouse. It should make quite a difference overall, especially on slower machines. LAME's performance impact should be minimal.
0 (Low priority)
Priority 0 assumes "IDLE" class, with delta 0.
LAME will have the lowest priority possible, and the encoding may be suspended very frequently by user interaction.
1 (Medium priority)
Priority 1 assumes "IDLE" class, with delta +31.
LAME won't interfere at all with what you're doing.
Recommended if you have a slower machine.
2 (Regular priority)
Priority 2 assumes "REGULAR" class, with delta -31.
LAME won't interfere with your activity. It'll run just like a regular process, but will spare just a bit of idle time for the system. Recommended for most users.
3 (High priority)
Priority 3 assumes "REGULAR" class, with delta 0.
LAME will run with a priority a bit higher than a normal process.
Good if you're just running LAME by itself or with moderate user interaction.
4 (Maximum priority)
Priority 4 assumes "REGULAR" class, with delta +31.
LAME will run with a very high priority, and may interfere with the machine response.
Recommended if you only intend to run LAME by itself, or if you have a fast processor.


Priority 1 or 2 is recommended for most users.



* -q 0..9 algorithm quality selection
Bitrate is of course the main influence on quality. The higher the bitrate, the higher the quality. But for a given bitrate, we have a choice of algorithms to determine the best scalefactors and huffman encoding (noise shaping).

-q 0: use slowest & best possible version of all algorithms. -q 0 and -q 1 are slow and may not produce significantly higher quality.

-q 2: recommended. Same as -h.

-q 5: default value. Good speed, reasonable quality.

-q 7: same as -f. Very fast, ok quality. (psycho acoustics are used for pre-echo & M/S, but no noise shaping is done.

-q 9: disables almost all algorithms including psy-model. poor quality.



* -r input file is raw pcm
Assume the input file is raw pcm. Sampling rate and mono/stereo/jstereo must be specified on the command line. Without -r, LAME will perform several fseek()'s on the input file looking for WAV and AIFF headers.
Might not be available on your release.



* --resample 8/11.025/12/16/22.05/24/32/44.1/48 output sampling frequency in kHz
Select ouptut sampling frequency (for encoding only).
If not specified, LAME will automatically resample the input when using high compression ratios.



* --r3mix r3mix VBR preset
Uses r3mix VBR preset.
See www.r3mix.net for more details.



* -s 8/11.025/12/16/22.05/24/32/44.1/48 sampling frequency
Required only for raw PCM input files. Otherwise it will be determined from the header of the input file.

LAME will automatically resample the input file to one of the supported MP3 samplerates if necessary.



* -S / --silent / --quiet silent operation
Don't print progress report.



* --scale n scales input by n
* --scale-l n scales input channel 0 (left) by n
* --scale-r n scales input channel 1 (right) by n
Scales input by n. This just multiplies the PCM data (after it has been converted to floating point) by n.

n > 1: increase volume
n = 1: no effect
n < 1: reduce volume

Use with care, since most MP3 decoders will truncate data which decodes to values greater than 32768.



* --short use short blocks
Let LAME use short blocks when appropriate. It is the default setting.




* --strictly-enforce-ISO strict ISO compliance
With this option, LAME will enforce the 7680 bit limitation on total frame size.
This results in many wasted bits for high bitrate encodings but will ensure strict ISO compatibility. This compatibility might be important for hardware players.




* -t disable INFO/WAV header
Disable writing of the INFO Tag on encoding.
This tag in embedded in frame 0 of the MP3 file. It includes some information about the encoding options of the file, and in VBR it lets VBR aware players correctly seek and compute playing times of VBR files.

When '--decode' is specified (decode to WAV), this flag will disable writing of the WAV header. The output will be raw pcm, native endian format. Use -x to swap bytes.



* -V 0...9 VBR quality setting
Enable VBR (Variable BitRate) and specifies the value of VBR quality.
default=4
0=highest quality.



* --vbr-new new VBR mode
Invokes the newest VBR algorithm. During the development of version 3.90, considerable tuning was done on this algorithm, and it is now considered to be on par with the original --vbr-old.
It has the added advantage of being very fast (over twice as fast as --vbr-old).



* --vbr-old older VBR mode
Invokes the oldest, most tested VBR algorithm. It produces very good quality files, though is not very fast. This has, up through v3.89, been considered the "workhorse" VBR algorithm.



* --verbose verbosity
Print a lot of information on screen.



* -x swapbytes
Swap bytes in the input file or ouptut file when using --decode.
For sorting out little endian/big endian type problems. If your encodings sounds like static, try this first.



* -X 0...7 change quality measure
When LAME searches for a "good" quantization, it has to compare the actual one with the best one found so far. The comparison says which one is better, the best so far or the actual. The -X parameter selects between different approaches to make this decision, -X0 beeing the default mode:

-X0
The criterions are (in order of importance):
* less distorted scalefactor bands
* the sum of noise over the thresholds is lower
* the total noise is lower

-X1
The actual is better if the maximum noise over all scalefactor bands is less than the best so far .

-X2
The actual is better if the total sum of noise is lower than the best so far.

-X3
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 best so far plus 2db.

-X4
Not yet documented.

-X5
The criterions are (in order of importance):
* the sum of noise over the thresholds is lower
* the total sum of noise is lower

-X6
The criterions are (in order of importance):
* the sum of noise over the thresholds is lower
* the maximum noise over all scalefactor bands is lower
* the total sum of noise is lower

-X7
The criterions are:
* less distorted scalefactor bands
or
* the sum of noise over the thresholds is lower

[zen]

ok thanks for the reply...I'll try and make some sense of it

[reddshack]

Zen,

If you are using the lame_enc.dll internal encoder, you must use the "modded" audiograbber to be able to input 'command line' style prompts. Go to http://www.audiograbber.com-us.net/b...ML/003517.html for instructions on how to use and obtain it.

The easiest way, however, would be to just use the external lame.exe encoder. This allows for all the command line inputs without modding.

Hope this helps!!

[zen]

all i want to do is rip at 128-192 vbr joint stereo...surely i don't need to mod anything to be able to do this?

also I just replaced the old exe file with the new one, and there appears to be no differance at all, I can't find the command stuff...

could someone please tell me how (preferably step by step) to set up audiograbber so I can rip it as above... I'm tempted to just rip at 128 and leave it at that without all the complications... please help!

[reddshack]

I don't use VBR so I'm clueless on that. I guess I should learn.

Go to the Audiograbber forum and search, read, or ask any question you have. I've read a few posts that explain VBR with lame and Audiograbber.

http://www.audiograbber.com-us.net/b...i/Ultimate.cgi

[Mike]

I looked into this and I don't think that Audiograbber, using an internal LAME encoder, easily lets you set a specific VBR range. (Maybe someone else with more Audiograbber experience will know otherwise.) CDex does, one of the reasons I use it; you may want to check it out.

With some experimenting on a single sample song, I came up with the following average bitrates at the indicated VBR levels, using Audiograbber.

VBR level - bitrate

0 - 205
1 - 193
2 - 187
3 - 183
4 - 177
5 - 165
6 - 147
7 - 134
8 - 120
9 - 119

This should give you what you need, at least relatively speaking; hope this helps--

[zen]

thanks for that, i just had a look on the audiograbber forum too, and found out that you can click on the little yellow vbr tabs to give you an average bitrate... this helps loads, but (and maybe i'm being picky) i'd still like to know that the maximum any of my files will go to is 192 and minimum is 128... perhaps there is a way somehow, i hope!

[Mike]

Yep--switch to CDex. (Really, it's not all that hard to learn, and you only do it once.) Maybe someone else will know another way ...

Interestingly, my average bitrates seemed to be 20-30 above what Audiograbber estimates the bitrates at each level to be.

[reddshack]

Interestingly, my average bitrates seemed to be 20-30 above what Audiograbber estimates the bitrates at each level to be.

That might have to do with the method of VBR you are using.

[zen]

I'd rather not switch to cdex, mostly because of speed reasons. I hear audiograbber is the fastest ripper out there, and because my computer is so old/crap i need all the help i can get!

also, in the 'Quality' part of the ripper, underneath the mono/joint stereo etc. part it has 'voice/low/normal/high' does anyone know what this means? I've been using high, but now with the space issue i'm going to choose normal... even though i don't know what it means! how does this affect the mp3?

*I just downloaded cdex and eac. cdex refuses to work (I just get the dreaded blue screen) and eac seems to be half made (no words on the tabs, half of the options aren't there...). i don't know what to do, all i want is fast ripper that does 128/192 vbr and so far the only ripper that does this is dbpoweramp (which is good but far too slow)*

[Mike]

From CDex help:

Quality (default value Normal):

With the LAME encoder, you can specify the output quality; thus you can trade off encoding time against sound quality. The default (normal) is recommended for the lower bitrates (<160 kbps), high quality for bitrates >160 kbps. The voice quality is more or less optimized to generate the best quality for voice. Details of the R3Mix quality option can be found at http://www.r3mix.net.

[Dunno]

You can specify the min and max bitrates with vbr but that requires a command line input with switches derived from that Long table above, an easier way to tweak mp3 is to use Razorlame, it has a GUI interface for LAME switches. the following command line
-b 128 -m j -V 1 -B 192 --lowpass 18 -q 2
was derived from Razorlames GUI, cut and paste it into the command line of AG. This set of swtiches will set the low bitrate to 128 and the highest to 192, but its variable between the two limits. (for a definition of the other switches refer table above), oh to get to the command line in AG, goto mp3 settings, tick External encoder, select Lamexxxx from drop down list, then in "Arguments" paste the command line from above, set extension to mp3, click OK and you're good to go. (easier than this ya don't get)

OR, if you use the default settings,
Extract below from AG's built in help file.
(Read "Quality" and "Bitrate slider" paragraphs below for your answers ).

Bitrate slider
Drag the slider to select the bitrate. 128 Kbit/s is the most common bitrate. You can see the actual bitrate or VBR mode on a label right above the slider. If you click that bitrate label it will expand and show the frequency for CBR modes and the average bitrate for VBR modes.

Quality
Here is a page that describes the different stereo modes.
Set to Voice if you record voice like a radio show or to low, normal or high quality. Low quality is the same as fast mode in previous versions of Audiograbber. Low quality is about twice as fast as Normal quality while high quality is around 30% slower. High quality is of course supposed to give the maximum sound quality for the bitrate.

P. S.
If you dont fancy the command line stuff, use AG's internal LAME encoder and manually set the VBR quality settings to the bit rate you desire, Adjustment via the VBR Bitrate Slider bar, ( see highligthed section above).

[zen]

ok thanks for your help guys...i'm getting the hang of it now!

although i've been comparing file sizes and 128/192 vbr joint stereo is a lot bigger that 128 cbr stereo...and with minimal sound difference (as far as I can hear....so....

I'm ripping at 128 stereo! I know im stupid and I wasted a lot of your time, but i do appreciate it, ive finally sorted how im going to rip my cds!

[reddshack]

I rip at 192 CBR quality High - Stereo......

Did you know that VBR ripping takes muuuuuuch longer than CBR Zen? Looks like you will be saving a lot of time with the CBR methoed you chose. Have fun with it all, it can take awhile (but in the end is sooo rewarding to have all your music)!!