There are various adpcm flavors. Which are you using specifically? A
Cool, game soundtrack ripping for the win. I may be able to calculate those parameters for codecs like that, but I guess it really depends on the average error of the encoders used in the first place... which would require that you either have the original encoders handy, or the original uncompressed audio for comparison.
foo_abx 1.3.4 reportfoobar2000 v1.1.162013/01/29 22:59:35File A: C:\Documents and Settings\Administrator.MISO-33497FE14F.000\My Documents\Med Sci 1 msadpcm.wavFile B: C:\Documents and Settings\Administrator.MISO-33497FE14F.000\My Documents\Med Sci 1 dvi.wav22:59:35 : Test started.23:00:14 : 01/01 50.0%23:00:23 : 02/02 25.0%23:00:34 : 03/03 12.5%23:00:43 : 04/04 6.3%23:01:01 : 05/05 3.1%23:01:23 : 06/06 1.6%23:01:32 : 07/07 0.8%23:01:51 : 08/08 0.4%23:02:06 : 09/09 0.2%23:02:12 : 10/10 0.1%23:02:22 : 11/11 0.0%23:02:31 : 12/12 0.0%23:02:33 : Test finished. ---------- Total: 12/12 (0.0%)
MS ADPCM has a far lower noise level in my opinion...
Well, I used the normalize function of soundforge .It says that the RMS for file "Eric_Brosius___02___Med_Sci_1___44_Khz_lossless.flac" (i.e 44khz, lossless) is -19db Now for the error , I compute the difference between the previous file and ADCPM version, (i.e Eric_Brosius___02___Med_Sci_1___44_Khz_IMA_ADPCM.wav)using the channel converter function of soundforge.The normalize function says that the RMS of error is -38.2 dbSo basically you have two number for 44hz version :signal: -19 dbnoise: -38.2 dbbut sorry I don't know the final formula to compute the snr, especially if you consider that soundforge give negative values for RMS .
Thank you very much, but what does the first value mean? RMS of what? The second I understand is of noise but the first...
ADPCM does not do any psychoacoustic tricks so if the SNR is about 38 dB it should sound as bad as 6-bit PCM but it doesn't.
When I'm thinking about it, what is the dynamic range of ADPCM anyways? Will it be 96 dB because of being decoded to 16-bit PCM, but with a low SNR, thus explaining the paradox
Thank you, but what does "signal: -19 db", ratio of signal to what? If the next values compute the RMS of noise.
Quote from: Neuron on 30 January, 2013, 08:14:47 AMThank you, but what does "signal: -19 db", ratio of signal to what? If the next values compute the RMS of noise.It's the value returned by the normalize function of soundforge.If you have -19 db for signal, and -38.2 db for noize, then you can deduce,that the signal is 19.2 db louder than the noize (38.2-19).I guess 0 db, would represent the loudest signal that the normalize function could measure on a file.Regarding 8 bit , well I decreased the bit depth of your file "Eric_Brosius___02___Med_Sci_1___44_Khz_lossless.flac"to 8 bit using the most straightforward algorithm (no noise shaping).Then again using most straightforward algorithm, I've converted it to 16 bit,and calculated difference with original file.RMS of difference measured , is -96 db, which could be interpreted that there's no meaningful (or no) difference.EDIT: I've done the same experience, with a random file, similar result, not sure what to conclude.
EDIT: I've done the same experience, with a random file, similar result, not sure what to conclude.EDIT 2: Listened to 8bit version of my random file from speakers, it doesn't sound bad to me. I won't do an ABX test though.
I got a "noise signal", peaking only 11.3 dB below the top.
Well, neither 4-bit ADPCM or 8-bit PCM are really "bad".
I did an Audacity convert to mono + match both volumes using Amplify + invert one of the track and Mix and render way of subtracting one track from the other. I got a "noise signal", peaking only 11.3 dB below the top. The weird thing is, the quieter parts of the song (not silence, only 9-10 dB below maximum) had "noise" as quiet as -45 dB while the louder parts went up to 11.3 dB. This is not all that is strange, because the "noise" left by the subtraction is actually a recognisable signal of the music, although very noisy (I will post a sample in the uploads section). By contrast, the dithered 8-bit signal is very audibly noisy, but it leaves a normal noise signal at -35.3 dB (which is a bit weird as well as 8-bit sound should have a 48 dB SNR and dithered 42 dB as I used only 1-bits of triangular dither, plus, the original signal is slightly present in the noise as well). The undithered 8-bit signal produces normal random noise at -48.2 dB as expected. Both dithered and undithered 8-bit signal have obvious, strong noise in them, unlike the ADPCM files.So, how to make sense of this? And why every difference signal except for the non-dithered 8-bit one has recognisable music it it?
The default signal compression encoding on a DS0 is either ?-law (mu-law) PCM (North America and Japan) or A-law PCM (Europe and most of the rest of the world). These are logarithmic compression systems where a 13 or 14 bit linear PCM sample number is mapped into an 8 bit value.
Quote from: Neuron on 30 January, 2013, 10:00:58 AMI got a "noise signal", peaking only 11.3 dB below the top.Measuring RMS is more representative of loudness than peak.QuoteWell, neither 4-bit ADPCM or 8-bit PCM are really "bad".I guess , that higher bit depth is only useful, if you want to reach higher dynamic range.