jump3r

Statement

A copy of ther copy of an unofficial LAME mp3 library port to Java from https://sourceforge.net/projects/jump3r/ by Hanns Holger Rutz

I (Vladimir Tsichevski) made the following changes to the project:

removed sbt build files
added Maven artefacts
to compile: mvn compile
to create self-contained jar: sbt assembly

Below are the original README and USAGE (adapted for markdown).

README

jump3r - Java Unofficial MP3 EncodeR

...is a Java version of lame-3.98.4

Java port created by Ken Händel.

Original sources by the authors of Lame:

http://www.sourceforge.net/projects/lame

USAGE

java -jar jump3r.jar [options] inputfile [outputfile]

For more options, just type:

java -jar jump3r.jar --help

Constant Bitrate Examples:

fixed bit rate jstereo 128 kbps encoding:

java -jar jump3r.jar sample.wav sample.mp3

fixed bit rate jstereo 128 kbps encoding, higher quality: (recommended)

java -jar jump3r.jar -h sample.wav sample.mp3

Fast encode, low quality (no noise shaping)

java -jar jump3r.jar -f sample.wav sample.mp3

Variable Bitrate Examples:

LAME has two types of variable bitrate: ABR and VBR.

ABR is the type of variable bitrate encoding usually found in other MP3 encoders, Vorbis and AAC. The number of bits is determined by some metric (like perceptual entropy, or just the number of bits needed for a certain set of encoding tables), and it is not based on computing the actual encoding/quantization error. ABR should always give results equal or better than CBR:

ABR: (--abr <x> means encode with an average bitrate of around x kbps)

java -jar jump3r.jar -h --abr 128  sample.wav sample.mp3

VBR is a true variable bitrate mode which bases the number of bits for each frame on the measured quantization error relative to the estimated allowed masking. There are 10 compression levels defined, ranging from 0=lowest compression to 9 highest compression. The resulting filesizes depend on the input material. On typical music you can expect -V 5 resulting in files averaging 132 kbps, -V 2 averaging 200 kbps.

Variable Bitrate (VBR): (use -V n to adjust quality/filesize)

java -jar jump3r.jar -V 2 sample.wav sample.mp3

LOW BITRATES

At lower bitrates, (like 24 kbps per channel), it is recommended that you use a 16 kHz sampling rate combined with lowpass filtering. LAME, as well as commercial encoders (FhG, Xing) will do this automatically. However, if you feel there is too much (or not enough) lowpass filtering, you may need to try different values of the lowpass cutoff and passband width (--resample, --lowpass and --lowpass-width options).

options guide:

These options are explained in detail below.

Quality related:

-m m/s/j/f/a : mode selection
-q n : Internal algorithm quality setting 0..9.
- 0 = slowest algorithms, but potentially highest quality
- 9 = faster algorithms, very poor quality
-h : same as -q2
-f : same as -q7

Constant Bit Rate (CBR)

-b n : set bitrate (8, 16, 24, ..., 320)
--freeformat : produce a free format bitstream. User must also specify a bitrate with -b, between 8 and 640 kbps.

Variable Bit Rate (VBR)

-v : VBR
--vbr-old : use old variable bitrate (VBR) routine
--vbr-new : use new variable bitrate (VBR) routine (default)
-V n: VBR quality setting (0=highest quality, 9=lowest)
-b n : specify a minimum allowed bitrate (8,16,24,...,320)
-B n : specify a maximum allowed bitrate (8,16,24,...,320)
-F : strictly enforce minimum bitrate
-t : disable VBR informational tag
--nohist: disable display of VBR bitrate histogram
--abr n : specify average bitrate desired

Operational:

-r : assume input file is raw PCM
-s n : input sampling frequency in kHz (for raw PCM input files)
--resample n: output sampling frequency
--mp3input : input file is an MP3 file. decode using mpglib/mpg123
-x : swap bytes of input file
--scale <arg> : multiply PCM input by
--scale-l <arg> : scale channel 0 (left) input (multiply PCM data) by
--scale-r <arg> : scale channel 1 (right) input (multiply PCM data) by
-a : downmix stereo input file to mono .mp3
-e n/5/c : de-emphasis
-p : add CRC error protection
-c : mark the encoded file as copyrighted
-o : mark the encoded file as a copy
-S : don't print progress report, VBR histogram
--strictly-enforce-ISO : comply as much as possible to ISO MPEG spec
--replaygain-fast : compute RG fast but slightly inaccurately (default)
--replaygain-accurate : compute RG more accurately and find the peak sample
--noreplaygain : disable ReplayGain analysis
--clipdetect : enable --replaygain-accurate and print a message whether clipping occurs and how far the waveform is from full scale
--decode : assume input file is an mp3 file, and decode to wav.
-t : disable writing of WAV header when using --decode (decode to raw pcm, native endian format (use -x to swap))

ID3 tagging:

--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 : same as --pad-id3v2-size 128
--pad-id3v2-size <num> : adds version 2 tag, pad with extra <num> 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.

Windows and OS/2-specific options:

--priority <type> sets the process priority

options not yet described:

--nores : disable bit reservoir
--disptime
--lowpass
--lowpass-width
--highpass
--highpass-width

Detailed description of all options in alphabetical order

downmix

-a

mix the stereo input file to mono and encode as mono.

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 java -jar jump3r.jar -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.

average bitrate encoding (aka Safe VBR)

--abr n

turns on encoding with a targeted average bitrate of n kbps, allowing to use frames of different sizes. The allowed range of n is 8...320 kbps, you can use any integer value within that range.

bitrate

-b  n

For MPEG-1 (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 MPEG-2 (sampling frequencies of 16, 22.05 and 24 kHz)
n = 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160

For MPEG-2.5 (sampling frequencies of 8, 11.025 and 12 kHz)
n = 8, 16, 24, 32, 40, 48, 56, 64

The bitrate to be used. Default is 128 kbps MPEG1, 80 kbps MPEG2.

When used with variable bitrate encodings (VBR), -b specifies the minimum bitrate to use. This is useful only if you need to circumvent a buggy hardware device with strange bitrate constrains.

max bitrate

-B  n

see also option -b for allowed bitrates.

Maximum allowed bitrate when using VBR/ABR.

Using -B is NOT RECOMMENDED. A 128 kbps CBR bitstream, because of the bit reservoir, can actually have frames which use as many bits as a 320 kbps frame. ABR/VBR modes minimize the use of the bit reservoir, and thus need to allow 320 kbps frames to get the same flexibility as CBR streams. This is useful only if you need to circumvent a buggy hardware device with strange bitrate constrains.

copyright

-c

mark the encoded file as copyrighted

clipping detection

--clipdetect

Enable --replaygain-accurate and print a message whether clipping occurs and how far in dB the waveform is from full scale.

mpglib decode capability

--decode

This just uses LAME's mpg123/mpglib interface to decode an MP3 file to a wav file. The input file can be any input type supported by encoding, including .mp3 (layers 1, 2 and 3).

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

de-emphasis

-e  n/5/c   

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.

fast mode

-f

Same as -q 7.

NOT RECOMMENDED. Use when encoding speed is critical and encoding quality does not matter. Disable noise shaping. Psycho acoustics are used only for bit allocation and pre-echo detection.

strictly enforce VBR minimum bitrate

-F

strictly enforce VBR minimum bitrate. With out this option, the minimum bitrate will be ignored for passages of analog silence.

free format bitstreams

--freeformat

LAME will produce a fixed bitrate, free format bitstream. User must specify the desired bitrate in kbps, which can be any integer between 8 and 640.

Not supported by most decoders. Compliant decoders (of which there are few) are only required to support up to 320 kbps.

Decoders which can handle free format:

                                     supports up to
MAD                      				640 kbps
jump3r									550 kbps  
Freeamp:                 				440 kbps
l3dec:                   				310 kbps

high quality

-h

use some quality improvements. The same as -q 2.

Modes

-m m           mono
-m s           stereo
-m j           joint stereo
-m f           forced mid/side stereo
-m d           dual (independent) channels
-m a           auto

MONO is the default mode for mono input files. If -m m is specified for a stereo input file, the two channels will be averaged into a mono signal.

STEREO

JOINT STEREO is the default mode for stereo files with fixed bitrates of 128 kbps or less. At higher fixed bitrates, the default is stereo. For VBR encoding, jstereo is the default for VBR_q >4, and stereo is the default for VBR_q <=4. You can override all of these defaults by specifying the mode on the command line.

jstereo means the encoder can use (on a frame by frame bases) either regular stereo (just encode left and right channels independently) or mid/side stereo. In mid/side stereo, the mid (L+R) and side (L-R) channels are encoded, and more bits are allocated to the mid channel than the side channel. This will effectively increase the bandwidth if the signal does not have too much stereo separation.

Mid/side stereo is basically a trick to increase bandwidth. At 128 kbps, it is clearly worth while. At higher bitrates it is less useful.

For truly mono content, use -m m, which will automatically down sample your input file to mono. This will produce 30% better results over -m j.

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.

FORCED MID/SIDE STEREO forces all frames to be encoded mid/side stereo. It should only be used if you are sure every frame of the input file has very little stereo seperation.

DUAL CHANNELS Not supported.

AUTO

Auto select should select (if input is stereo)

      8 kbps   Mono
 16- 96 kbps   Intensity Stereo (if available, otherwise Joint Stereo)
112-128 kbps   Joint Stereo -mj
160-192 kbps   -mj with variable mid/side threshold
224-320 kbps   Independent Stereo -ms

MP3 input file

--mp3input

Assume the input file is a MP3 file. LAME will decode the input file before re-encoding it. Since MP3 is a lossy format, this is not recommended in general. But it is useful for creating low bitrate mp3s from high bitrate mp3s. If the filename ends in .mp3 LAME will assume it is an MP3.

disable histogram display

--nohist

By default, LAME will display a bitrate histogram while producing VBR mp3 files. This will disable that feature.

disable ReplayGain analysis

--noreplaygain

By default ReplayGain analysis is enabled. This switch disables it.

See also: --replaygain-accurate, --replaygain-fast

non-original

-o

mark the encoded file as a copy

CRC error protection

-p

turn on CRC error protection.
Yes this really does work correctly in LAME. However, it takes 16 bits per frame that would otherwise be used for encoding.

algorithm quality selection

-q n

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 scale factors and huffman encoding (noise shaping).

-q 0 : use slowest and best possible version of all algorithms.
-q 2 : recommended. Same as -h. -q 0 and -q 1 are slow and may not produce significantly higher quality.
-q 5 : default value. Good speed, reasonable quality
-q 7 : same as -f. Very fast, ok quality. (psycho acoustics are used for pre-echo and M/S, but no noise shaping is done.
-q 9 : disables almost all algorithms including psy-model. poor quality.

input file is raw pcm

-r

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.

slightly more accurate ReplayGain analysis and finding the peak sample

--replaygain-accurate

Enable decoding on the fly. Compute "Radio" ReplayGain on the decoded data stream. Find the peak sample of the decoded data stream and store it in the file.

ReplayGain analysis does not affect the content of a compressed data stream itself, 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 http://www.replaygain.org/

By default, LAME performs ReplayGain analysis on the input data (after the user-specified volume scaling). This behaviour might give slightly inaccurate results because the data on the output of a lossy compression/decompression sequence differs from the initial input data. When --replaygain-accurate is specified the mp3 stream gets decoded on the fly and the analysis is performed on the decoded data stream. Although theoretically this method gives more accurate results, it has several disadvantages:

tests have shown that the difference between the ReplayGain values computed on the input data and decoded data is usually no greater than 0.5dB, although the minimum volume difference the human ear can perceive is about 1.0dB
decoding on the fly significantly slows down the encoding process

The apparent advantage is that:

with --replaygain-accurate the peak sample is determined and stored in the file. The knowledge of the peak sample can be useful to decoders (players) to prevent a negative effect called 'clipping' that introduces distortion into sound.

Only the "Radio" ReplayGain value is computed. It is stored in the LAME tag. The analysis is performed with the reference volume equal to 89dB. Note: the reference volume has been changed from 83dB on transition from version 3.95 to 3.95.1.

See also: --replaygain-fast, --noreplaygain, --clipdetect

fast ReplayGain analysis

--replaygain-fast

Compute "Radio" ReplayGain of the input data stream after user-specified volume scaling and/or resampling.

This switch is enabled by default.

See also: --replaygain-accurate, --noreplaygain

output sampling frequency in kHz

--resample  n

where n = 8, 11.025, 12, 16, 22.05, 24, 32, 44.1, 48

Output sampling frequency. Resample the input if necessary.

If not specified, LAME may sometimes resample automatically when faced with extreme compression conditions (like encoding a 44.1 kHz input file at 32 kbps). To disable this automatic resampling, you have to use --resamle to set the output sample-rate equal to the input sample-rate. In that case, LAME will not perform any extra computations.

sampling frequency in kHz

-s  n

where n = sampling rate in kHz.

Required for raw PCM input files. Otherwise it will be determined from the header information in the input file.

LAME will automatically resample the input file to one of the supported MP3 sample-rates if necessary.

silent operation

-S

don't print progress report

scale

--scale <arg>

Scales input by <arg>. This just multiplies the PCM data (after it has been converted to floating point) by <arg>.

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

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

strict ISO complience

--strictly-enforce-ISO

With this option, LAME will enforce the 7680 bit limitation on total frame size. This results in many wasted bits for high bitrate encodings.

disable VBR tag

-t

Disable writing of the VBR Tag (only valid if -v flag is specified) This tag in embedded in frame 0 of the MP3 file. It lets VBR aware players correctly seek and compute playing times of VBR files.

When --decode is specified (decode mp3 to wav), this flag will disable writing the WAV header. The output will be raw pcm, native endian format. Use -x to swap bytes.

variable bit rate (VBR)

-v

Turn on VBR. There are several ways you can use VBR. I personally like using VBR to get files slightly bigger than 128 kbps files, where the extra bits are used for the occasional difficult-to-encode frame. For this, try specifying a minimum bitrate to use with VBR:

java -jar jump3r.jar -v      -b 112  input.wav output.mp3

If the file is too big, use -V n, where n = 0...9

java -jar jump3r.jar -v -V n -b 112  input.wav output.mp3

If you want to use VBR to get the maximum compression possible, and for this, you can try:

java -jar jump3r.jar -v  input.wav output.mp3
java -jar jump3r.jar -v -V n input.wav output.mp3         (to vary quality/filesize)

VBR quality setting

-V n

n = 0...9. Specifies the value of VBR_q. default = 4, highest quality = 0, smallest files = 9

Using -V 6 or higher (lower quality) is NOT RECOMMENDED.
ABR will produce better results.

How is VBR_q used?

The value of VBR_q influences two basic parameters of LAME's psycho acoustics:

the absolute threshold of hearing
the sample to noise ratio

The lower the VBR_q value the lower the injected quantization noise will be.

NOTE No psy-model is perfect, so there can often be distortion which is audible even though the psy-model claims it is not! Thus using a small minimum bitrate can result in some aggressive compression and audible distortion even with -V 0. Thus using -V 0 does not sound better than a fixed 256 kbps encoding. For example: suppose in the 1 kHz frequency band the psy-model claims 20 dB of distortion will not be detectable by the human ear, so LAME VBR-0 will compress that frequency band as much as possible and introduce at most 20 dB of distortion. Using a fixed 256 kbps framesize, LAME could end up introducing only 2 dB of distortion. If the psy-model was correct, they will both sound the same. If the psy-model was wrong, the VBR-0 result can sound worse.

swapbytes

-x

swap bytes in the input file (and output file when using --decode). For sorting out little endian/big endian type problems. If your encodings sound like static, try this first.

tsichevski/jump3r