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
Disable the bit reservoir
Enforce ISO bitstream
De-emphasis
n = (none, default)
5 = 0/15 microseconds
c = citt j.17
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:
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
Auto
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
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
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.
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
Lowpass filter width
Highpass filter
Highpass filter width
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.
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
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
ATH auto adjustment type
ATH auto adjustment sensitivity
Activation offset in -/+ dB for ATH auto adjustment.
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
Disable temporal masking effect
M/S switching criterion
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.
Sets the process priority:
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.
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...
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
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/ |