Creare un mp3 di qualità help!

se poi vuoi usare un decoder esterno (ergo non una DLL, ma un eseguibile da riga di comando e senza interfaccia grafica) supponiamo ad esempio il LAME.EXE in una delle sue numerose versioni devi conoscere un pò di nozioni:


LAME version 3.95 MMX (http://www.mp3dev.org/)

usage: lame [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)
--nogaptags allow the use of VBR tags in gapless encoding

Operational options:
-m <mode> (s)tereo, (j)oint, (f)orce, (m)ono
default is (s) or (j) depending on bitrate
force = force ms_stereo on all frames.
-a downmix from stereo to mono file for mono encoding
--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>
--replaygain-accurate slightly more accurate and much slower ReplayGain
--clipdetect determine whether clipping occurs and display an appropriate

message. store the MP3 peak sample in LAME Tag
--preset type type must be "medium", "standard", "extreme", "insane",
or a value for an average desired bitrate and depending
on the value specified, appropriate quality settings will
be used.
"--preset help" gives more info on these


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
--brief print more useful information
--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
--cbr enforce use of constant bitrate

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
-T enable and force 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 [0-4]
--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
--nssafejoint M/S switching criterion
--nsmsfix <arg> M/S switching tuning [effective 0-3.5]
--interch x adjust inter-channel masking ratio
--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
--shortthreshold x,y short block switching threshold, x for L/R/M channel,
y for S channel
Noise Shaping related:
--substep n use pseudo substep noise shaping method types 0-2

experimental switches:
-X n[,m] 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 [n] toggles the scalefac-scale and subblock gain feature on
if n is set and minus, only scalefac-scale is enabled


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
--ignore-tag-errors ignore errors in values passed for tags

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.


MS-Windows-specific options:
--priority <type> sets the process priority:
0,1 = Low priority (IDLE_PRIORITY_CLASS)
2 = normal priority (NORMAL_PRIORITY_CLASS, defau
lt)
3,4 = High priority (HIGH_PRIORITY_CLASS))
Note: Calling '--priority' without a parameter will select priority 0.


Platform specific:
--noasm <instructions> disable assembly optimizations for mmx/3dnow/sse



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



direi che non è il caso di complicarsi la vita, no?

[pholcus]

http://www.hydrogenaudio.org/index.p...ST&f=15&t=203&

Qui ci sono i preset..usate questi..

--alt-preset standard (~190 kbit/s, typical 180 ... 220)

--alt-preset fast standard (~190 kbit/s, faster but potentially lower quality)

--alt-preset extreme (~250 kbit/s, typical 220 ... 270)

--alt-preset fast extreme (~250 kbit/s, faster but potentially lower quality)

--alt-preset insane (320 kbit/s CBR, highest possible quality)

avrei una domanda:

che differenza c'e' tra Stereo e Joint-Stereo?

quale e' il migliore?

ciao

[antonyb]

Originariamente inviato da davlak
scusa ma che ti frega delle opzioni da riga di comando che ci vuole un trattato per impararle (se vuoi te le passo pure ma sono in inglese e dopo non mi chiedere la traduzione )

Allora l'interfaccia grafica che ce l'ha a fare Audiograbber?

usa queste opzioni nelle impostazioni MP3

si hai ragione non conviene complicarsi la vita!

sono stato nelle opzioni mp3 ma nella codifica interna non ho il lame, come mai?

[antonyb]

Originariamente inviato da afterhours
avrei una domanda:

che differenza c'e' tra Stereo e Joint-Stereo?

quale e' il migliore?

ciao

...non lo so bene ma credo sia meglio stereo, j stereo da quello che ho capito unisce le frequenze.

bisognerebbe chiedere a davlak che lo vedo molto esperto!

che ne dici dav?

[antonyb]

Originariamente inviato da pholcus
http://www.hydrogenaudio.org/index.p...ST&f=15&t=203&

Qui ci sono i preset..usate questi..

--alt-preset standard (~190 kbit/s, typical 180 ... 220)

--alt-preset fast standard (~190 kbit/s, faster but potentially lower quality)

--alt-preset extreme (~250 kbit/s, typical 220 ... 270)

--alt-preset fast extreme (~250 kbit/s, faster but potentially lower quality)

--alt-preset insane (320 kbit/s CBR, highest possible quality)

...molti preset creano file j stereo bo non so se vadano bene!!!

@after dall'help di audiograbber:
Different stereo modes

A common question is what the different stereo modes is used for in MP3 files.


· Joint Stereo: Joint Stereo shares certain bits between high frequency left and right channels. This improves compression efficiency at a slight loss of stereo separation. Lower frequencies are treated as normal stereo. Use Joint Stereo to obtain the best overall quality at mid-to-lower bit rates. < 224 Kbit.

· Stereo: Stereo includes two independent channels. The total bit rate remains constant, but the split between the channels can vary. The Encoder uses this flexibility to improve quality by allocating more bits to the channel with the more dynamic signal. Use the Stereo setting for best quality stereo audio at higher bit rates.

· Dual Stereo: Dual Stereo includes two completely independent channels (left/right), each with half the total bit rate. In effect, it is two mono files packed into a single file. Dual is generally used for multi-lingual audio programs.

The higher bitrate the larger file and better sound quality. 128 Kbit is MP3 standard and 64 Kbit is WMA standard. 128 Kbit is the same as 16000 bytes (128000 / 8). That is 16 Kbytes per second or just a little bit less than one MB per minute of music. That is equal to a compression ratio of 1:11.025. 128 Kbit is usually referred to as cd quality. That might be true for most songs but some songs needs higher bitrates to sound perfect. Use headphones to listen for artifacts in the MP3 or WMA files.

@antonyb

devi scompattare da questo il lame_enc.dll (vers. 3.96) nella cartella audiograbber

http://lkc.uw.hu/letoltesek/windows/lame/lame-3.96.zip

Grazie

[antonyb]

Originariamente inviato da davlak
@after dall'help di audiograbber:
Different stereo modes

A common question is what the different stereo modes is used for in MP3 files.


· Joint Stereo: Joint Stereo shares certain bits between high frequency left and right channels. This improves compression efficiency at a slight loss of stereo separation. Lower frequencies are treated as normal stereo. Use Joint Stereo to obtain the best overall quality at mid-to-lower bit rates. < 224 Kbit.

· Stereo: Stereo includes two independent channels. The total bit rate remains constant, but the split between the channels can vary. The Encoder uses this flexibility to improve quality by allocating more bits to the channel with the more dynamic signal. Use the Stereo setting for best quality stereo audio at higher bit rates.

· Dual Stereo: Dual Stereo includes two completely independent channels (left/right), each with half the total bit rate. In effect, it is two mono files packed into a single file. Dual is generally used for multi-lingual audio programs.

The higher bitrate the larger file and better sound quality. 128 Kbit is MP3 standard and 64 Kbit is WMA standard. 128 Kbit is the same as 16000 bytes (128000 / 8). That is 16 Kbytes per second or just a little bit less than one MB per minute of music. That is equal to a compression ratio of 1:11.025. 128 Kbit is usually referred to as cd quality. That might be true for most songs but some songs needs higher bitrates to sound perfect. Use headphones to listen for artifacts in the MP3 or WMA files.

@antonyb

devi scompattare da questo il lame_enc.dll (vers. 3.96) nella cartella audiograbber

http://lkc.uw.hu/letoltesek/windows/lame/lame-3.96.zip

Ok ...va bene dav mi metto a lavoro e ti faccio sapere presto!

molte grazie per ora!

[antonyb]

....si vengono un po grandi ma ne vale la pena si sentono molto bene, domani passo da un mio amico che ha delle cuffie molto buone così sento meglio la differenza!

grazie dav!

Più grandi si, ma non di molto; ho calcolato che con le impostazioni che ti ho suggerito (quelle che uso anche io) c'è un aumento del 30% circa rispetto al classico 192 kbps, ma vale davvero la pena.
Quando avrai fatto "la mano" con i vari formati in mp3 ti spiegherò come estrarli dalle immagini di CD audio montate su unità virtuali (usando Alcohol o le Daemon Tools), e non direttamente dal CD...e lì la differenza diventa davvero sensibile.
Per adesso buon ascolto

[antonyb]

Originariamente inviato da davlak
Più grandi si, ma non di molto; ho calcolato che con le impostazioni che ti ho suggerito (quelle che uso anche io) c'è un aumento del 30% circa rispetto al classico 192 kbps, ma vale davvero la pena.
Quando avrai fatto "la mano" con i vari formati in mp3 ti spiegherò come estrarli dalle immagini di CD audio montate su unità virtuali (usando Alcohol o le Daemon Tools), e non direttamente dal CD...e lì la differenza diventa davvero sensibile.
Per adesso buon ascolto

.....facciamo finta che gia sono esperto dav!
come funge quello che dici tu??

1) che sw di masterizzazione usi?
2) hai mai usato le daemon tools?
3) puoi installare alcohol 120% (che è senza'altro il miglior sw per questo tipo di lavoro)?

[antonyb]

Originariamente inviato da davlak
1) che sw di masterizzazione usi?
2) hai mai usato le daemon tools?
3) puoi installare alcohol 120% (che è senza'altro il miglior sw per questo tipo di lavoro)?

masterizzo con clone oppure con nero e a volte con disc juggler.

non ho mai usato daemon tools non so neanche che fa!

Ok installo subito alcohol 120%

............chissà che hai in mente di fare!