BLADE ENC
This is a plugin offered from the program's web site as 'optional ' . I have also installed it  for testing
Blade  is one of the oldest freeware  converters but with closed coding  although recently ity has been open source . The version used is 0.94 




As you may see the  codec configuration  is very simple . Just the bitrate and some basic  adjustments with also separate channel encoding (the dual channel)
Bitrates  are standard  with 32, 48 , 56 , 64 , 80-, 96 , 128 192  224 256 and 320 being the highest

Here are the measurements  with BLAde


         CD         WAV        b/W
320      14.19      12.9      22kHz   6.87
192      15,87!!    13.12     22 kHz  6.62
128      17.81!!    16.06     16 kHz /peak 800 Hz
96       18.53      16.75     -7 db<2.4k ,-4<16k,-42>16
48       18.46      16.77     -40<2.6, -17<-5, 7.5-16> -7.5


The first  column is ripping  time from CD of a song of 2,26 mins
Secoind  columns from the WAV ripped  song
b/w  is the  measured bandwitdth  in terms of db and kHz in plain  text . though it can sound too technical  will be analsyzed a bit below.

MY comments:
As shown above  the  ripping times are slightly increased as far as the bitrates are  lowered! For me its is quite incommon but it is posibly the  problem with the coder
For the bandwidth measurements , 192 and 320 are OK  as a  'store format'
128 means  Hi quality FM format with a peak of the tone LA
96  : everything below the speaker's voice is truncated  leaving just the trebles to be heard . testing  one song  converted from 128 to 96 , the sound was  quite 'plastic'
48 :  the spectrogram showed a stair shaped drawing , meaning that  the 'higher tones  have higher audio level  ' . in plain language it means that there is no bass and the trebles have a bad mood. A  song cannot be heard  properly in this mode

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OGG VORBIS

The ogg vorbis  format  , though i have rread somewhere is much more xcomplex thanthe standard MP3 format seems  here to be in a oversimlified adjustments
As is quite known , ogg vorbis offers  quite better audio then MP3 (at least for the FhG fastenc codec )  thogh  from my measuremenrs below  the  new LAME  goes a little better !



and



As shown in the two  ictures above the  adjustmsnt are very simple and seem tobe  very clse to the LAME  format

And here are my measurementsga

OGG  from the above  WAV saved file of 2.26 mins
A192     1.20.30
A126     1.30.66
A065     1.30.38

V0.6    18.74!!!
v0.4    19.59
v0.2    20.53
v0!     20,53
v-1.9   21.25

as in  result , the average bitrates require neartly the half time of the song, whnile the variable bit rates  are quite fast and increase  slightly with the decrease of the variation level

And here are the  bandwidth  results  from various bitrates qith the seocnd way of processing: WAV file > Ogg then ogg> wav out for spectral analysis on cooledit


A 192  22k   linear
A 126  18K1  slightly curved up as above 8 kHzby 3db
A 64   15k   curved after 6Kto 0 on 15 kHz
--
V0.6   22k   linear
v0.3   17k   slightly curved up as above 8 kHzby 3db
v0.1   16k
v0     15k2  curved after 6Kto 0 on 15 kHz  (as A64)
v-0.9! 15    curved after 6Kto 0 on 15 kHz,rippled 11- 15 (A48 same )


From the abovee  the  latest vorbis encoder ofers slighly less  spectrum then the lame on the  highest fidelity !
But from the resulted adio even on the 45 /v-0.9  is very clear in contrast to the same bitrate of LAme that had artifacts

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FAAC  the freeware converter  for AAC and MP4
In the reality   AAC and MP4 are exactly the same!




As you wil see there  is noting  special with this  converter .
It used  a CBR  rating per channel  though the file sizes  are always the same  for 8 - 48 kbps  in my tests. It did happena also to converrt files of dferent sizes
IN conrtast  the VBR levels  shown a difenrence in the file sizes  for the 1 min noise bering 1.42MB  for the 100% dropping 967KB  (as 128 for MP3 ) for the 42% and 448KB (64kb MP3 size) for the 10%  with obvious  audio artifacts
   


And here is nothing special except for the object type . Thw main and low options  are very fast , but the LTP  (Long Term Prediction) is very slow , upto 4 times

For more info on this format you may look at : http://wiki.hydrogenaudio.org/index.php?title=AAC

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FLAC

Ands another  codec with lossless AAC  The amin site for FLAC describes it this way :




As shwon above , the FLAC converter uses several presets  and  two stereo modes  the standard joint stereo and the non standard 'adaptive'
The  presets  are between  the fastest  and the  best compression in 8 steps  total Sizes come between 9.76 and 9.84 MB for the preset 8 and 1 . The times  to convert  a 4 min  noise WAV file are:
19 secs for the best compression  or 4.75 sec per MB  resulting to 40.942.070 Bytes
7.93 secs per MB  for the  festest or 1.98 secs  resulting to 41287186 BYtes
 Original WAV noise file  is 42336044 BYtes

Another experiment :
A song of 3.55
-18.15  secs   27815488 BYtes   for the  highest compression
-7.34   secs    30719266  BYtes  for the fastest compression




here is a definaition for the subset  on http://flac.sourceforge.net/format.html





And here the most intereting point is :
  apodization , which means taperring or trimming. A little more googling shows that apodization is clearing the  residual parts of the bel curve berlow a predefind level More here: http://en.wikipedia.org/wiki/Apodization_function which sown  also various functions

Info on the  linear predictor:

IN this and the following articles you will find som measurenemts  for the encoders used with bonk:
first let me advise you the  method  i used for  measuring

1. For the times measurements : Used
- A song from a CD  of 2.26 mins ripped directly . I order no to damage the CD i then  ripped it to WAV  format
-a stop watch unit from  DS clock,  a  simple visual  (and free )clock  program , with .01 sec accuracy. most of the results have +/- 0.15 secs tolerance 
- Snagit one of the oldest capturing programs  for capturing part of the screen
- After snagit was started capturing   i start the stop watch and then bonkenc
- After Bonkenc ended its task , i stopped the timer and then the capturing
- i then measure the bars progressing and do the time  subraction

Here is a video  with the above methodology

>>>
2. For the  sprectrum measurememnts

A WAV file  is first made with  white noise  of one minute duration on 44000 Hz  and 16 bit 
-then is converted  in all formats , compression levels and quality adjustments and named accordinly (LAM/98/5 for example )
.Though Audacity is one solution for making the WAV file , Cooledit (a very old version) is still  my most preferable for allthe next measurements.

Then :
For the mp3s : - the file is dragged into Cooledit marked for about 15 -20 secs then scanned from the spectrum analysis window   for the RMS values via a relatively  big sampling rate ie 2048 or 4096 and 'calculating the curve into text'
For files  not supported  by Cooledit :
-conversion  via Bonk to WAV
-draging into Cooledit  and then using the above  method

Though  i do not think  that this method is flawy , if soemone has any claims  plase let me know


And  a short description of my computer system:
Pentium PC with  3 GHZ
768 MB
80GB HArd disk

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LAME (Lame Ain't an MP3 Encoder!!!)
For me Lame  is the most interesting converter that has a ton of adjustments  best for the very geeky , a much  higher level than myself who is mostly  to the audio response level .
http://wiki.hydrogenaudio.org/index.php?title=LAME




The lame encoder  has too many parameters to use . the most important for me is the quality item that can convert from  just  4 secs for the worst quality to 25 secs  for the better quality. IN this paramter  i prever to use the highest possible number - usually 3 and 2 -  inorder to have the best  audio quality - though mine ears  do not  find any  significant diference .

There are also several presets as medium (VBR), standard , extreme, insane , adn their  fast versions R3mix and ABR All except ABR do not uncover their presets. ABR standsas for automatic bitrate ans is shown in the VBR tab
  However  the  documentataion reveals this  info:
 -preset medium  > V4 rh
 -preset fast medium > v4 mrh
 -preset  standard > v2rh
 -preset  fast standard >v2mrh
- preset extreme v0
- preseet fast extremev 0
- preset insane C320


The bitrate can be from 8 to 320  to preset numbers ( 8, 16, 24, 32 , 40 48 56 64 80 96 112 128 144 160 192 224 256 and 320)
There  is also a size ratio.   in uncomon numbers as 90!   the resulted  file is deleted  immediately after the  conversion.

The quality rate  is soemthing for me very interesting . Quality  zero means here the best  quality (therefore the highest time for encoding ) and 9 the lowest quality (and very fast )

Beofre going on  with the other  tabs let me show my results in ripping /converting from CD and WAV onto MP3: or song of 2.26 min (146 sec)

at CBR on 128 kb
       CD           wav
 at 0 > 1.13.67    1.12.72   ie the half !!!
    1 > 0.39.50      39.00   nearly the half of 0
    2 > 0.33.06      30.12
    3 > 0.19.06      17,13
    4 > 0.18.94      16,63
    5 > 0.17.81      15,28
    6 > 0.17.19      14,63
    7 > 0.12.13       9,82
    8 > 0.12.19      10,16
    9 > 0.08.46       5,29
 LAME COMPRESSIONS on 128kbps

Here is per compression rate at CBR times in secs
320 kb  17.06   14.1
256     16,63   14.17
192     16,25   13,84/13.66
128     17,91   15.47
96      18,75   16.41
48      16.44   14,5
--
24       9,35   7.16  requires adj to output sampling rate
8        8.47   6,38 requires adj to output sampling rate

the two latest formats  require  downsampling (audio processng )in order  to perform the coversion , usaually on 8 or 11 kHz

And here is a test based graphical  analysis of the most common CBR rates:


256   20 kHz

192   19kHz
practically  this bandwidth (B/W) means a store level

128-9 22kHz!!

128-5 19

128-0 19 kHz
the worst quality level due to speed offers a  artificial whole b/w level . the other quality levels offer a Cd quality

112/9 22kHz !!
112/5 19 kHz
112/0 19 kHz
the same as above

96/9  22 kHz  distorted!
96/5  19 kHz
96-9  19 kHz curved down by 3 db


64-0  8k @-3 max to 18 kHz
64-5  18 k linear
64-9  22k! linear
and here is the funny: the best quality  cuts freqs above the 8 kHz though the wirrst modes ofer a 'transparent' quality

Lame 48 0          10 kHz  
Lame 48 5          17 khxz
Lame 48 9          22 kHz!!


16-9   6 kHz   @-3 8 khzoff
16-5   5 k ,6k @-3 8 K off
16-0   1.7kHz  @-3 4.5 kHz coff


once again the  numers show narroweer audio for the best encoding !!
==
ENCODER



And above LAME shows  its more important characteristics the  two variable rates . for a  better explanation i lend the info from the wiki  of LAME  as foun in the site of Hydrogen audio:


CBR:  the standard constant bit rate

CBR is only recommended for usage in streaming situations where the upper bitrate must be strictly enforced.

VBR: this is the variable bit rate

VBR is best used to target a specific quality level, instead of a specific bitrate. The final file size of a VBR encode is less predictable than with ABR, but the quality is usually better.

Unlike other MP3 encoders which do VBR encoding based on predictions of output quality, LAME's default VBR method tests the actual output quality to ensure the desired quality level is always achieved.

rh adn mrth are two dierent algorithms.Mrth  offers abou two times faster processing

ABR: average bitrate mode.
AS shown above in the photo , VBR can be asjusted  in tbetween minimums and maximums
misc


The misc tab determines some usual MP3 standards as copy right bits rtc  but en/disables padding on frames
ISO compliance : This refers  mostly  for the MP3 harware players  , in order  for the resulted file  to be compaliant to the MP3 standard (compatibility problems)
padding : a patch


expert tab:


thisis possibly the most dificult setting  with the  explanation from other pages :

ATH : absolute threshold of hearing 


Temporal masking effect
       



Audio processing



This pane is the easiest for me and the simplest  comparing to other more 'advanced' adjustments !
There are the standard resample rates of 8 ,11 22 44 and 48 kHz , the first two necessary  for the lower bit ratets  to operate.

Enabling  filtering can make you remove some frequencies  or determine a whole bandwidth to cut
For better  undersitanding it is better   to experiment a bit with them . Notice that lowpass filter  must be bigger number than the low pass frequency !!

Here is a oversimplified listing  of the MP3 compressions :Notice that as the number in the left pane is higher ,  the  higher the filter and thus higher fidelity audio. The higher numbers can offer the abilities or the  lower numbers  too.  
Notice: The program  used is Music match jukebox that offers  the  curves shown in http://zlgr.multiply.com/reviews/item/22

kbps	approximate usage -
8 - 11 kbps	suitable for recording telephonic conversations , voice  recording . narrowband AM radio
16	the above plus high fidelity  speech, AM radio, low fidelity streaming radio
24- 32kbps	high quality speech , AM radio [medium and shortwaves ]
44-56	low fidelity FM radio
64	older fidelity FM radio
96-128	CD  quality audio
192	CD quality -
256	'Store'  mode for CD  quality

and for WMA - fior anyone interested , as  tested with Musicmatch Jukebox . For me  mostly

5-8-12	just simple voice recording
16-22	high quality speech recording , 22 seems  as HIFI
44	good for FM
48	CD audio  compromise , best for mass storage
64	CD  recording   = 128 on MP3
96	'high quality' CD recording =192 on MP3

And notice the  following:
The above listing is still controversial  and depends on the program and codecs you use. Older  lame codec [used in goldwave ] does not offer the same audio quality
and several  comments on windows media wma files:
- offering lower dynamic range :  they can clip very low voices
- on hardware players :
   - they can drain the double current .
   - in most older MP3 players i tested , the WMA codec  can  offer better audio than its MP3  codec||
   - the  wma codec may introduce its own noise  with 'cracks' in sudden audio changes  or in loud points.

There were several bad links .Please read it again as the pictures are now corectly shown

--------- Given that many users on Multiply rip or covert their music in quite big files of over 7.5 MB fr a 4 minute song ,i decided to write this article to explain why it is not necessary to use very high sampling rates.

At same time I will notice to you why I use the WMA format in 48 kbps to include mass material into a small MP3 player as shown in my previous reviews

It is important to advise you that I was in the audio technology and HiFi's since 1980 , including also the MP3 tech form ca 1997 and my first ever MP3 player was with CD on 2000.Also i used several professional equipment the older days and I know very much about it.

This is a abridged version of my blog in http://zlgr.multiply.com/journal/item/41 with a little more explanation in the pictorials If someone is still very boring to read this whole message ,please check the flies on the http://zlgr.multiply.com/journal/item/55 for his personal evaluation , though it is very worth of reading

And here is the basic background for the above theory .

1.Make a file with white noise . This noise is used primarily for testing purposes in audio laboratories. it can be used to measure the linearity of audio systems including the audio part of radio transmitters. The easiest way to make this white noise is with the Cooledit program.

For best measurements i made a file of 60 secs making a file of 10M336B [10336 Kb]I also deliberately use the linear frequency response in order to find more easily the cutoff frequencies than with the standard logarithmic freq response

This is our original WAV file's frequency response , designed from a white noise of 0 db level. As you may notice it is a totally linear, and will be used as reference to the results from various compressions shown later

2.Convert the files toMP3 and /or WMA in the most popular conversions . I choose the MP3 format on 128 [standard]192 [broadcast mode ] and 256kb [store mode] while for WMA the 44 [FM] ,. 48[I say just CD] 64 [default near CD ]and 96 [CD ] kb.

I always used the Musicmatch Jukebox in its 6th version which is very light comparing to the newest versions and has very good results especially if converting from CDs and MPs to WMAs . And though Cooledit has MP3 save option

3.View the resulted files back to Coledit's frequency analysis screen

:

	256 kb . As you will see the 256 kb sampling rate offers a very high audio bandwidth of ca 21 kHz filtering out everything above this frequency ,with more than 60 db cut off .But these cut off frequencies are inaudible to the ear .. Tthe file size is 1876 KBytes
	192kb The curve here goes to the 20 kHz , and about 60 db cutoff above these frequencies. This is the maximum the ear can listen The file size is 1407 Kb
	128 kb And here is the standard MP3 with strong cutoff above 15.900 Hz . I do not see any significant difference between 128 and 192 in my ears though still 112 with a cut off at 14000 and 96 kb at 12000 sound nearly the same The file size is 938 kb on the 128kb

AS Cooledit cannot open WMA files , i used first a third party program to convert the WMA files iinto WAV and then opened with Cooledit , and here are the results :

48 kbps	48 64 and 96 kb Here all the WMA files have the same audio bandwidth at 20 kHz with the only difference on the dynamic level. ,
64 kbps	The dynamic level is ca 36db , for the 48 kb , ca 40 db for the 64 kb and ca 55 db for the 96 kb compression. The 40 db are the audio levels used by most music groups. More than 60 db are used only by classical music And the file sizes are 367,484 and 718 kB respectively . IN practice that means that all these modes can offfer just high quality audio though very low audio levels are 'removed' .
96 kbps	In a very careful listening you can see some minor artifacts on the 48 kbps especially in drums and high trebled audio.
WMA 44	And here a funny case of the 44 kb WMA. Instead of having a smooth cuve , that reminds a triple 'treble control' , a part of the bandwidth is somehow cut out However this happens with the MMJB 6 . Using another converter as for example the MP3WM OGG converter, the results are also diferent . The results in the MP3 conversion showed me a flat bandwidth for 128 to 256 to 22000 with 96kb having -6db on 16000max .while the WMA compressions rates were showed as a bandstop filter analogous to the compression rate ...

Furthermore these are the results with the over mentioned program . I mean , using a different program to convert files gives a different result
AS for example Cooleditwith its MP3 plugin ca offer linear response to the maximum of the bandwidth.

At another example i was also using the ACE mult iformat converter . The results once opened or converted to WAV (for WMA) have made several interesting results . Please see the picture below, it is an animation!

From the above results , it is clear that ACE can technically offer better audio than MMJB on the MP3 format , but worse than MMJB in te WMA format

You  can find on the enclosed PDF file