MP3 vs. MPC vs. Ogg: Low volume test Options

[tigre]

Hi there!

Inspired by Using DirectSound SSRC Plugin thread I wanted to know what's the difference between outputs of foobar 2000 and winamp/other players. I figured out following test:

1. Take a sample (I did it with 30 seconds of music, could be done with test signals too), convert it to 32 bit resolution in Cool Edit Pro.

2. Apply logarithmic fadeout to the whole sample: 0 to -150 db (using CEP).

3. Save or dither to 24bit -> save, depending on the needs of lossy encoders (Step 4).

4. Encode with lame 3.90.2 -api, mpc 1.15i braindead, vorbis 1.0 -q 10 (this should not be the weak link).

5. Playback encoded files in tested players and capture output to wav file, resolution 24bit/96kHz if possible. The problem here is that I don't know any tool for this, I asked for advice in this thread. Because of this I've only tested this on foobar2000 0.5 beta so far, as its diskwriting feature is capable of 24bit/96kHz output. So until now it's not a player output comparison but a encoder comparison as you'll see soon.

6. Aply logarithmic fadein 0 -> +150db to the whole sample with CEP to bring the volume back to original level everywhere.

7. Listening to the resulting files.

The reason for this proceeding was to "exaggerate" the differences that occur at very low volume and make them audible.

My observations: All files started to contain aditional noise compared to the original after ~1/2 of playing time. At this point started what surprised me: The mp3 sounded like the original + increasing noise till the end of the sample while MPC and Vorbis started with slight artifacts and sounded more and more awfull: ogg somehow like underwater, mpc like NUMLock's lame audiophile preset and worse.

So is mp3 superior at low volumes? Not really. I gave mp3 an advantage by using CBR. A quick test using lame ape (=VBR) showed the same awfulness as Vorbis and mpc. Besides, I noticed that with decreasing volume the bitrate of all 3 codes decreased too.

Conclusion? I'm not sure, but maybe there's something that could be improved (adaptive ath model for high resolution signals, or ...). Imagine, you have a high resolution source (DVD audio ...) and music that uses its huge dynamic range (not completely of course) - and equipment with really high SNR ... this could be an issue.

I hope I haven't just wased my time and space on this forum . Thoughts welcome.

[tigre]

@NumLock:
You're right. I did another quick test. This time more "realistic": I took my test sample (very familiar with it now), and "enveloped" (=amplified dynamically) it after changing resolution to 32bit in the following way:
0-5sec: 0dB; 5-10sec.: fadout 0 -> -40 db; 10-30sec: fadout -40 -> -60dB; 30-35sec.: fadein -60 -> 0dB; 35sec-end: 0dB
I encoded the resulting sample (dithered to 16bit) with --preset extreme and tried to ABX (at a volume sounding comfortable at the beginning and at the end of the sample. I couldn't hear/ABX any differences. Focusing on the artifacts I heard in the test before was not successful, my impression was that if there'd have been artifacts, they were covered by noise (which is caused mainly by my hardware/environment, as "digital silence noise" sounds almost exactly the same at this volume to me.)

My conclusion: On music that has a wide dynamic range (50dB and more) - with slow volume changes, giving the ear/brain enough time to adapt - listened to with low noise equipment/environment (maybe from 24bit source) there could still be a audible difference to lossy encoded (especially mp3 as I haven't tested the other formats that accurately). Although I've not been able to prove (=ABX) it with my equipment/environment/abilities, I believe this is a resonable conclusion from my tests. If my reasoning is wrong, please convince me of the opposite. I'm open to this.