MPC Encoder Functions - Revision history

178.95.132.4: /* Quality oriented encoder functions */ fixing formatting again...

2016-10-19T08:26:37Z

Quality oriented encoder functions: fixing formatting again...

← Older revision		Revision as of 08:26, 19 October 2016
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	; --ltq_max x ''(x can be: -99 to 99; recommended: 60 to 99)''		; --ltq_max x ''(x can be: -99 to 99; recommended: 60 to 99)''
	: Maximum level for ltq, in [[dB]]. default is 83. - --ltq_gain x (x can be: -99 to 99; recommended: -12 to 5)		: Maximum level for ltq, in [[dB]]. default is 83.


			; --ltq_gain x ''(x can be: -99 to 99; recommended: -12 to 5)''
	: Adds offset of x db to chosen ltq. If you use negative number, you can make the hearing curve more sensitive (for more sensitive hearing), but it increases [[bitrate]]. If you use positive number, less bits will be needed.		: Adds offset of x db to chosen ltq. If you use negative number, you can make the hearing curve more sensitive (for more sensitive hearing), but it increases [[bitrate]]. If you use positive number, less bits will be needed.

178.95.132.4: /* Quality oriented encoder functions */ better formating

2016-10-19T08:16:55Z

Quality oriented encoder functions: better formating

← Older revision		Revision as of 08:16, 19 October 2016
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	==Quality oriented encoder functions==		==Quality oriented encoder functions==

	; --ms x (x can be: 0, 1, 2)		; --ms x ''(x can be: 0, 1, 2)''
	: Sets [[Mid-side stereo]] mode (channel coupling): 0 (off), 1 (on) or 2 (enhanced): 0 means that there's no channel coupling, 2 means channel coupling is more cautious. 1 means, channel coupling is not so cautious which may result joint stereo artifacts.		: Sets [[Mid-side stereo]] mode (channel coupling): 0 (off), 1 (on) or 2 (enhanced): 0 means that there's no channel coupling, 2 means channel coupling is more cautious. 1 means, channel coupling is not so cautious which may result joint stereo artifacts.

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	; --cvd x (x can be: 0, 1)		; --cvd x ''(x can be: 0, 1)''
	: Sets clearvoicedetection either off (0) or on (1 default).		: Sets clearvoicedetection either off (0) or on (1 default).
	: CVD is able to detect voice-like signals to give a higher quality with voices or sounds with harmonic spectra. It uses special analysis to detect [[harmonics]] with varying base [[frequency]] - the "normal" psychoacoustics are not able to detect such signals and will add audible noise to these signals.		: CVD is able to detect voice-like signals to give a higher quality with voices or sounds with harmonic spectra. It uses special analysis to detect [[harmonics]] with varying base [[frequency]] - the "normal" psychoacoustics are not able to detect such signals and will add audible noise to these signals.


	; --bw x (x can be: 0 to 22500) : Defines the max frequency [[bandwidth]] which can be encoded (actual frequency response depends also on [[LTQ\|ATM]]/[[ATH]]). Basically acts like a [[lowpass\|lowpass filter]].		; --bw x ''(x can be: 0 to 22500)''
			: Defines the max frequency [[bandwidth]] which can be encoded (actual frequency response depends also on [[LTQ\|ATM]]/[[ATH]]). Basically acts like a [[lowpass\|lowpass filter]].


	; --ltq x (x can be: iso, ank, fil) : Level threshold in quiet (also called as ATH) is a threshold or hearing curve. This is the sound pressure level (spl in db) below which the human hearing of most people is unable to perceive a sine-tone.		; --ltq x ''(x can be: iso, ank, fil)''
			: Level threshold in quiet (also called as ATH) is a threshold or hearing curve. This is the sound pressure level (spl in db) below which the human hearing of most people is unable to perceive a sine-tone.


	; --ltq_max x (x can be: -99 to 99; recommended: 60 to 99)		; --ltq_max x ''(x can be: -99 to 99; recommended: 60 to 99)''
	: ~~maximum~~ level for ltq, in [[dB]]. default is 83. - --ltq_gain x (x can be: -99 to 99; recommended: -12 to 5)		: Maximum level for ltq, in [[dB]]. default is 83. - --ltq_gain x (x can be: -99 to 99; recommended: -12 to 5)
	: Adds offset of x db to chosen ltq. If you use negative number, you can make the hearing curve more sensitive (for more sensitive hearing), but it increases [[bitrate]]. If you use positive number, less bits will be needed.		: Adds offset of x db to chosen ltq. If you use negative number, you can make the hearing curve more sensitive (for more sensitive hearing), but it increases [[bitrate]]. If you use positive number, less bits will be needed.


	; --ltq_var x (x can be: 0, 1)		; --ltq_var x ''(x can be: 0, 1)''
	: Adaptive threshold in quiet. default is 1.		: Adaptive threshold in quiet. default is 1.


	; --minSMR x (x can be: 0 to 3; recommended: 0)		; --minSMR x ''(x can be: 0 to 3; recommended: 0)''
	: Sets the minimum smr (signal to mask ratio) over full BandWidth. The higher the smr the higher the quality and bitrate. Setting -minSMR over 0 db will result in full BandWidth encoding, like in insane profile.		: Sets the minimum smr (signal to mask ratio) over full BandWidth. The higher the smr the higher the quality and bitrate. Setting -minSMR over 0 db will result in full BandWidth encoding, like in insane profile.


	; --tmpmask x (x can be: 0, 1)		; --tmpmask x ''(x can be: 0, 1)''
	: Sets post [[masking]] on or off. Temporal postmasking saves a few kbit/s because the human hearing has to "relax" after a sound event, so that the encoder can put a bit more distortion to the signal during this time (saves bits).		: Sets post [[masking]] on or off. Temporal postmasking saves a few kbit/s because the human hearing has to "relax" after a sound event, so that the encoder can put a bit more distortion to the signal during this time (saves bits).


	; --nmt x (x can be: 0 to 99; recommended: 6 to 16)		; --nmt x ''(x can be: 0 to 99; recommended: 6 to 16)''
	: Sets minimum smr (signal to mask ratio) for pure noisy sound. MPC encoder calculates a masking threshold. Noisy sound has high masking ratio. [[Subband]] coder like MPC works by adding noise ([[quantization]] error) to each subband. You can increase the quantization resolution (less quantization noise, higher [[bitrate]]) by raising smr.		: Sets minimum smr (signal to mask ratio) for pure noisy sound. MPC encoder calculates a masking threshold. Noisy sound has high masking ratio. [[Subband]] coder like MPC works by adding noise ([[quantization]] error) to each subband. You can increase the quantization resolution (less quantization noise, higher [[bitrate]]) by raising smr.

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	; --tmn x (x can be: 0 to 99; recommended: 22 to 32)		; --tmn x ''(x can be: 0 to 99; recommended: 22 to 32)''
	: Sets minimum smr for pure sinusoidal sound. Sinusoidal sound is very tonal (not noisy). This means that it does not have much masking capability. Quantization resolution (bitrate) must be high enough so that tonal sound is encoded without audible noise. Of course in normal music there is both noisy and tonal sound, so the masking threshold will be calculated accordingly. Also different resolutions of quantization can be assigned to the different frequency regions.		: Sets minimum smr for pure sinusoidal sound. Sinusoidal sound is very tonal (not noisy). This means that it does not have much masking capability. Quantization resolution (bitrate) must be high enough so that tonal sound is encoded without audible noise. Of course in normal music there is both noisy and tonal sound, so the masking threshold will be calculated accordingly. Also different resolutions of quantization can be assigned to the different frequency regions.


	; --ans x (x can be: 0 to 5; recommended: 5)		; --ans x ''(x can be: 0 to 5; recommended: 5)''
	: Adaptive noise shaping order. 0 means off, 1 to 5 means on. Default is 5.		: Adaptive noise shaping order. 0 means off, 1 to 5 means on. Default is 5.


	;--shortthr x (x can be for example: 5)		;--shortthr x ''(x can be for example: 5)''
	: Short fft threshold. default is 5.		: Short fft threshold. default is 5.


	; --transdet x (x can be for example: 100)		; --transdet x ''(x can be for example: 100)''
	: ~~slewrate~~ for [[transient]] detection. default is 100.		: Slewrate for [[transient]] detection. default is 100.


	; --minval x (x can be: 0, 1, 2)		; --minval x ''(x can be: 0, 1, 2)''
	: Method for calculating minval. 0: old method, 1: Buschmann method, 2: Klemm method.		: Method for calculating minval. 0: old method, 1: Buschmann method, 2: Klemm method.

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	; --scale x (x can be: 0.00 to 1.00; recommended 0.850 to 1.00)		; --scale x ''(x can be: 0.00 to 1.00; recommended 0.850 to 1.00)''
	: Method to overcome [[clipping]] of the encoded file. In the event that clipping occurs, the encoder (mppenc) will display a warning message and recommend a numerical value to process the audio so that clipping may cease. The numerical value is a percentage to the extent that the audio will be processed (1.0 = 100%, 0.85 = 85%). "--scale 1.00" means no additional clipping processing will be done - this is the desired situation.		: Method to overcome [[clipping]] of the encoded file. In the event that clipping occurs, the encoder (mppenc) will display a warning message and recommend a numerical value to process the audio so that clipping may cease. The numerical value is a percentage to the extent that the audio will be processed (1.0 = 100%, 0.85 = 85%). "--scale 1.00" means no additional clipping processing will be done - this is the desired situation.


	; --fadeshape x (x can be: 1 to 99; recommended: 1 to 10)		; --fadeshape x ''(x can be: 1 to 99; recommended: 1 to 10)''
	: Sets the fading scheme used. "Small values are first fast fading then slow fading. Large values are the opposite." - FrankKlemm?		: Sets the fading scheme used. "Small values are first fast fading then slow fading. Large values are the opposite." - FrankKlemm?


	; --fadein x / --fadeout x (x can be: 0 to 99; recommended: 0 to 5)		; --fadein x / --fadeout x ''(x can be: 0 to 99; recommended: 0 to 5)''
	: The encoder (mppenc) offers simple audio processing via fading. "X" indicates the number of seconds to which the encoder will fade the music. Useful for encoding live recordings when the whole recording will not be used (e.g. tracklisting, playlists, seamless listening, shuffled music, etc.).		: The encoder (mppenc) offers simple audio processing via fading. "X" indicates the number of seconds to which the encoder will fade the music. Useful for encoding live recordings when the whole recording will not be used (e.g. tracklisting, playlists, seamless listening, shuffled music, etc.).


	; --start x / --skip x (x can be: 0 to 99; recommended: 0 to 5)		; --start x / --skip x ''(x can be: 0 to 99; recommended: 0 to 5)''
	: Sets the number of seconds that the encoder will NOT PROCESS. (i.e. "--start 4" will skip the first 4 seconds of the source file and the encoder will begin encoding on the fourth second). "Start" and "skip" do the same thing.		: Sets the number of seconds that the encoder will NOT PROCESS. (i.e. "--start 4" will skip the first 4 seconds of the source file and the encoder will begin encoding on the fourth second). "Start" and "skip" do the same thing.

Rjamorim: delinkage

2005-07-13T16:21:27Z

delinkage

← Older revision		Revision as of 16:21, 13 July 2005
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	: Sets [[Mid-side stereo]] mode (channel coupling): 0 (off), 1 (on) or 2 (enhanced): 0 means that there's no channel coupling, 2 means channel coupling is more cautious. 1 means, channel coupling is not so cautious which may result joint stereo artifacts.		: Sets [[Mid-side stereo]] mode (channel coupling): 0 (off), 1 (on) or 2 (enhanced): 0 means that there's no channel coupling, 2 means channel coupling is more cautious. 1 means, channel coupling is not so cautious which may result joint stereo artifacts.

	: M/S-coding calculates a "mid"-channel by addition of left and right channel (l+r)/2 and a "side"-channel (l-r)/2. With more mono-like signals one can use less [[bitrate]] to encode the side-channel, so that the overall~~- [[~~bitrate]] will be less than encoding the left and right channel. If the psychoacoustics work well, there is no audible difference between m/s- coded or l/r-coded files. Mid/Side coding in MPC is [[subband]] selective, broadband 0-22khz is divided into 32 ~~[[subband\|~~subbands]]. [[Psychoacoustic\|Psychoacoustics]] calculates for each subband if mid/side coding should be used or not. This is different than in [[MP3]] encoding, where the full frame will be either m/s coded or l/r (true stereo) coded, so [[MP3]] mid/side coding is more likely to cause audible artifacts, unless tweaked to be very cautious (like nssafejoint in lame encoder).		: M/S-coding calculates a "mid"-channel by addition of left and right channel (l+r)/2 and a "side"-channel (l-r)/2. With more mono-like signals one can use less [[bitrate]] to encode the side-channel, so that the overall bitrate will be less than encoding the left and right channel. If the psychoacoustics work well, there is no audible difference between m/s- coded or l/r-coded files. Mid/Side coding in MPC is [[subband]] selective, broadband 0-22khz is divided into 32 subbands. [[Psychoacoustic\|Psychoacoustics]] calculates for each subband if mid/side coding should be used or not. This is different than in [[MP3]] encoding, where the full frame will be either m/s coded or l/r (true stereo) coded, so MP3 mid/side coding is more likely to cause audible artifacts, unless tweaked to be very cautious (like nssafejoint in lame encoder).


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	; --bw x (x can be: 0 to 22500) : Defines the max [[frequency]] [[bandwidth]] which can be encoded (actual frequency response depends also on [[LTQ\|ATM]]/[[ATH]]). Basically acts like a [[lowpass\|lowpass filter]].		; --bw x (x can be: 0 to 22500) : Defines the max frequency [[bandwidth]] which can be encoded (actual frequency response depends also on [[LTQ\|ATM]]/[[ATH]]). Basically acts like a [[lowpass\|lowpass filter]].


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	; --minSMR x (x can be: 0 to 3; recommended: 0)		; --minSMR x (x can be: 0 to 3; recommended: 0)
	: Sets the minimum smr (signal to mask ratio) over full BandWidth. The higher the smr the higher the quality and ~~BitRate~~. Setting -minSMR over 0 db will result in full BandWidth encoding, like in insane profile.		: Sets the minimum smr (signal to mask ratio) over full BandWidth. The higher the smr the higher the quality and bitrate. Setting -minSMR over 0 db will result in full BandWidth encoding, like in insane profile.


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	; --tmn x (x can be: 0 to 99; recommended: 22 to 32)		; --tmn x (x can be: 0 to 99; recommended: 22 to 32)
	: Sets minimum smr for pure sinusoidal sound. Sinusoidal sound is very tonal (not noisy). This means that it does not have much masking capability. [[Quantization]] resolution ([[bitrate]]) must be high enough so that tonal sound is encoded without audible noise. Of course in normal music there is both noisy and tonal sound, so the [[masking]] threshold will be calculated accordingly. Also different resolutions of quantization can be assigned to the different [[frequency]] regions.		: Sets minimum smr for pure sinusoidal sound. Sinusoidal sound is very tonal (not noisy). This means that it does not have much masking capability. Quantization resolution (bitrate) must be high enough so that tonal sound is encoded without audible noise. Of course in normal music there is both noisy and tonal sound, so the masking threshold will be calculated accordingly. Also different resolutions of quantization can be assigned to the different frequency regions.

Mix de Rax: Copied from Rarewares

2004-10-23T12:09:55Z

Copied from Rarewares

@@ Line 1: / Line 1: @@
-usage:
+==Quality oriented encoder functions==
-Standard options:
+; --ms x (x can be: 0, 1, 2)
-  --silent       do not write any message to the console (dflt: off)
+: Sets [[Mid-side stereo]] mode (channel coupling): 0 (off), 1 (on) or 2 (enhanced): 0 means that there's no channel coupling, 2 means channel coupling is more cautious. 1 means, channel coupling is not so cautious which may result joint stereo artifacts.
-Profile Options (Quality Presets):
+: M/S-coding calculates a "mid"-channel by addition of left and right channel (l+r)/2 and a "side"-channel (l-r)/2. With more mono-like signals one can use less [[bitrate]] to encode the side-channel, so that the overall- [[bitrate]] will be less than encoding the left and right channel. If the psychoacoustics work well, there is no audible difference between m/s- coded or l/r-coded files. Mid/Side coding in MPC is [[subband]] selective, broadband 0-22khz is divided into 32 [[subband|subbands]]. [[Psychoacoustic|Psychoacoustics]] calculates for each subband if mid/side coding should be used or not. This is different than in [[MP3]] encoding, where the full frame will be either m/s coded or l/r (true stereo) coded, so [[MP3]] mid/side coding is more likely to cause audible artifacts, unless tweaked to be very cautious (like nssafejoint in lame encoder).
-Examples:
-  mppenc inputfile.wav
+; --cvd x (x can be: 0, 1)
-  mppenc inputfile.wav outputfile.mpc
+: Sets clearvoicedetection either off (0) or on (1 default).
-  mppenc --xtreme inputfile.pac outputfile.mpc
+: CVD is able to detect voice-like signals to give a higher quality with voices or sounds with harmonic spectra. It uses special analysis to detect [[harmonics]] with varying base [[frequency]] - the "normal" psychoacoustics are not able to detect such signals and will add audible noise to these signals.
-  mppenc --silent --radio inputfile.pac outputfile.mpc

Mix de Rax at 14:11, 19 October 2004

2004-10-19T14:11:40Z

New page

usage:
mppenc [--options] <Input_File>
mppenc [--options] <Input_File> <Output_File>

Standard options:
--silent do not write any message to the console (dflt: off)
--deleteinput delete input file after encoding (dflt: off)
--overwrite overwrite existing destination file (dflt: off)
--fade sec fade in and out with 'sec' duration (dflt: 0.0)

Profile Options (Quality Presets):
--thumb low quality/internet, (typ. 58... 86 kbps)
--radio medium (MP3) quality, (typ. 112...152 kbps)
--standard high quality (dflt), (typ. 142...184 kbps)
--xtreme extreme high quality, (typ. 168...212 kbps)

Examples:
mppenc inputfile.wav
mppenc inputfile.wav outputfile.mpc
mppenc --xtreme inputfile.pac outputfile.mpc
mppenc --silent --radio inputfile.pac outputfile.mpc