Ustawienia LAME / Tryb Ekspert / Ustawienia zaawansowane

Zakładka "Ustawenia LAME" (Tryb Ekspert / Ustawienia zaawansowane)

Bitstream
ReplayGain
MP3 Header
Raw PCM
Filters
ATH (Absolute Threshold of Hearing)
PSY related
LAME priority
Assembly optimizations
Other options
Experimental switches
Dodatkowe parametry LAME

Bitstream

Disable the bit reservoir

Each frame will then become independent from previous ones, but the quality will be lower.

Enforce ISO bitstream

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.

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.

ReplayGain

Szybki (domyślny)

Compute ReplayGain fast but slightly inaccurately (default).

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

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.

Only the "RadioGain" 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.

This switch is enabled by default.

Dokładny

Compute ReplayGain more accurately and find the peak sample.

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 behavior 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 "RadioGain" 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.

This option is not usable if the MP3 decoder was explicitly disabled in the build of LAME. (Note: if LAME is compiled without the MP3 decoder, ReplayGain analysis is performed on the input data after user-specified volume scaling).

Wyłącz

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

Auto

Stosowany jest domyślny ReplayGain. (w wersji LAME 3.98 - Szybki)

MP3 Header

Ochrona CRC

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.

Copyright

Mark the encoded file as copyrighted.

Oryginał

Mark the encoded file as original.

Raw PCM

Input file is a raw PCM

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

Input sampling frequency

8/11.025/12/16/22.05/24/32/44.1/48
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.

Input bit width

8/16/24/32
Required only for raw PCM input files. Otherwise it will be determined from the header of the input file.

Downmix to mono

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.

Filters

LAME default

Use default filters.

Disable all filters

Tells the encoder to use full bandwidth and to disable all filters. By default, the encoder uses some lowpass filtering at lower 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 filter

Set a lowpass filtering frequency. Frequencies above the specified one will be cutoff.

Lowpass filter width

Set the width of the lowpass filter. The default value is 15% of the lowpass frequency.

Highpass filter

Set an highpass filtering frequency. Frequencies below the specified one will be cutoff.

Highpass filter width

Set the width of the highpass filter. The default value is 15% of the highpass frequency.

Output sampling frequency

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

ATH (Absolute Threshold of Hearing)

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.

ATH only for short blocks

Ignore psychoacoustic model for short blocks, use ATH only.

Only use ATH

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.

ATH type

0/1/2

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.

Lowers ATH by

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 useful.

ATH auto adjustment type

ATH auto adjust: 0 - no adjustment, 1-3 - loudnes based adjustment.

ATH auto adjustment sensitivity

Activation offset in -/+ dB for ATH auto adjustment.

PSY related

Do not use short blocks

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

Use short blocks when appriopriate

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

Use only short blocks

Use only short blocks, no long ones.

Disable temporal masking effect

Don't make use of the temporal masking effect.

M/S switching criterion

--nssafejoint

M/S switching tuning

Effective 0-3.5.

Inter-channel masking ratio

Adjust inter-channel masking ratio.

ns-bass

Adjust masking for sfbs 0 - 6 (long) 0 - 5 (short).

ns-alto

Adjust masking for sfbs 7 - 13 (long) 6 - 10 (short).

ns-treble

Adjust masking for sfbs 14 - 21 (long) 11 - 12 (short).

ns-sfb21

Change ns-treble by x dB for sfb21.

Short block switching threshold...

Short block switching threshold, x for L/R/M channel, y for S channel.

LAME priority

Sets the process priority:

  • 0,1 = Low priority (IDLE_PRIORITY_CLASS)
  • 2 = normal priority (NORMAL_PRIORITY_CLASS, default)
  • 3,4 = High priority (HIGH_PRIORITY_CLASS)

Assembly optimizations

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.

Other options

Disable writing WAV header

When decoding to WAV, this option will disable writing of the WAV header. The output will be raw PCM, native endian format. Use -x to swap bytes.

Swap bytes...

Swap bytes in the input file or output file when decoding to WAV.
For sorting out little endian/big endian type problems. If your encodings sounds like static, try this first.

Pseudo substep...

Use pseudo substep noise shaping method types (0-2).

Experimental switches

X

Selects between different noise measurements, n for long block, m for short. If m is omitted, m = n.

Y

Lets LAME ignore noise in sfb21, like in CBR.

Z

Currently no effects.

Dodatkowe parametry LAME

W dolnej części tej zakładki znajduje się pole tekstowe, w którym można wpisać dodatkowe parametry LAME.

Jeśli zaznaczona zostanie opcja Używaj TYLKO poniższych parametrów, używane będą tylko parametry podane w poniższym polu tekstowym, a wszelkie dodatkowe ustawienia będą ignorowane!

INDEX http://www.pazera-software.com/products/lame-front-end/