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.
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
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 !
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
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
Ands another codec with lossless AAC The amin site for FLAC describes it this way :
FLAC stands for Free Lossless Audio Codec, an audio format similar to MP3, but lossless, meaning that audio is compressed in FLAC without any loss in quality. This is similar to how Zip works, except with FLAC you will get much better compression because it is designed specifically for audio, and you can play back compressed FLAC files in your favorite player (or your car or home stereo, see supported devices) just like you would an MP3 file.
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
FLAC specifies a subset of itself as the Subset format. The purpose of this is to ensure that any streams encoded according to the Subset are truly "streamable", meaning that a decoder that cannot seek within the stream can still pick up in the middle of the stream and start decoding.
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:
FLAC uses a class of computationally-efficient fixed linear predictors (for a good description, see audiopak and shorten). FLAC adds a fourth-order predictor to the zero-to-third-order predictors used by Shorten. Since the predictors are fixed, the predictor order is the only parameter that needs to be stored in the compressed stream. The error signal is then passed to the residual coder.