well, soundexpert.info does use a questionable testing technique based on strong assumptions that are very likely to be invalid.
I'm trying to assess whether I'll regret adding on to my music library using WMA 192 CBR files from here on out
QuoteI'm trying to assess whether I'll regret adding on to my music library using WMA 192 CBR files from here on outI've come to the conclusion that with lossy compression, (relative) ignorance really is bliss. By all means, do an ABX test to find out if you can spot blatantly bad quality, but the last thing you want is spend months training yourself to hear the most minute artefacts (that will be in there somewhere, even at 320 kbps), which will then ruin your listening experience forever.Similarly, if you train yourself to spot JPG compression artefacts in digital photos, the joy of browsing through pictures on Flickr (or whatever) will never be the same.
Quote from: SebastianG on 27 November, 2006, 05:00:52 PMwell, soundexpert.info does use a questionable testing technique based on strong assumptions that are very likely to be invalid.But isn't it also true that these "assumptions" will never result in a sample that sounds better than it would if the "assumptions" weren't in place?
After reading the PDF about their testing methodology (http://www.soundexpert.info/articles/IGIS.pdf) I fail to see the purpose of these tests. If I understood thing correctly, SoundExpert relies on techniques as "Sound artifacts amplification" in other words, comparing input and output signals and amplifying artifacts that are not perceived under normal conditions. What is the point of such tests?
Precisely why you've never seen me participate in any listening tests. Although I must confess that my aged ears probably wouldn't be much help anyway!!
After reading the PDF about their testing methodology (http://www.soundexpert.info/articles/IGIS.pdf) I fail to see the purpose of these tests. If I understood thing correctly, SoundExpert relies on techniques as "Sound artifacts amplification" in other words, comparing input and output signals and amplifying artifacts that are not perceived under normal conditions. What is the point of such tests?If I'm mistaken, please feel free to correct me...
Anyway, what I've determined is this contention that transcoding reduced the quality is WAY overblown (mostly by audio enthusiasts who are very critical as opposed to practical.)
So my conclusion is if you have high-quality source material (maybe the minimum might be AAC 128, MP3 160 VBR / 192 CBR, etc. etc.) then I think any loss of sound quality when you transcode to an equal or better encoding method is transparent for 99% of us.
I was recently involved in a debate over this very topic. I think transparency of a second encode is not only based on the settings but also on how large of a margin of transparency there is on the first encode. If the first encode employed the minimum settings to achieve transparency (and hence had a small margin), the second encode could easily not be transparent with an equal encoding method. You can easily prove this to yourself through blind testing.
I figured since I was tossing my CD's it was better to go with a very high quality encoding
...It is still not clear to me how it is possible to define a metric above transparency. ...
...What we're out for with lossy codecs is transparency, but transparency is a soft subject. It depends on the listener, and it depends upon the listener being trained to carefully listen.As for this it makes sense to provide for a test that makes it easy to hear artefacts....So to me any listening test is valuable, but in a restricted sense, and this holds for the soundexpert tests as well.
... all I'm wondering is where in the scale of quality does WMA 192 CBR fall? Is it better than the AAC 128kbit that iTunes puts out or about the same. Is it about the same as LAME MP3 Preset:Medium?
… such analytically computed ratings are usually located above 5th grade on impairment scale. This could be interpreted as quality margin or quality headroom of an audio device because the artifacts are beyond the threshold of human audibility. You may ask what the purpose of the margin is if sound artifacts are inaudible already. There are at least four reasons why this is important:• In general case perceived audio quality of a device/technology depends on sound samples used for testing. Theoretically a listening test has to be performed with infinite number of sound samples in order to prove for sure that tested device will not produce unexpected “surprises” on real-world audio material. In practice a limited set of typical or problem (“killer”) sound samples is used. Then testing results are just generalized on all audio. Obviously quality margin makes that generalization more grounded and lessens the probability of getting artifacts on audio material not used during the test. • Very often audio devices/technologies are used in chains – connected one after another. In most cases this accumulates sound degradation throughout the chain. Quality margin of each device is highly desirable to lessen overall distortion level. • Such post processors as equalizers, spatializers, SRS and many others usually reveal sound artifacts inaudible in “normal” cases. Some quality headroom helps to use all those popular sound enhancements safely without danger of discovering drawbacks of other audio components.• Human hearing abilities differ from person to person. Averaged results of any listening test have to be applied with great caution to someone’s particular situation especially if that someone has “golden ears”. Such person needs audio equipment with sufficient quality margin in order to be satisfied. Sound quality margin is not something completely new. Well known technical audio parameter – THD is used quite similarly: measured on pure sine wave and corresponding to perceived audio quality not very well it have to be as low as possible – far beyond human abilities to hear such low distortions of pure waves.
Obviously quality margin makes that generalization more grounded and lessens the probability of getting artifacts on audio material not used during the test.
I still don't understand how can you define a metric "above transparency" for a psychoacoustic codec.
Also I really think that the THD example can give a wrong information to the reader, since it is defined without regarding listener's perception.
Last, while it is very difficult to write a codec that outperform the others in "traditional" abx tests, is it hard to write one that exploits soundexpert's tests at a given bitrate? (even simply preprocessing the audio signal and then using a standard codec)
I use to think that the quality of the second encode depends on how the algorithms of the two encoders "stack", so I always prefer to ABX.It is still not clear to me how it is possible to define a metric above transparency. Soundexperts methodology sounds strange to me because I can't imagine a "neutral" way to amplify distortions of psychoacoustic algorithms, and why this way could be indicative of the behavior of the coded file to different types of further processing... but I'm sure these were discussed here and I can't really add anything to the discussion
Sound artifacts amplification for hearing is quite the same as magnifying glass for vision.
A more appropriate analogy would be...
You try to artificially increase percibility of artifacts for understandable reasons. But you're doing it via amplifying the difference which is just a very very bad idea. What's a "correct amplification" in the above example? You could try to detect the movement and extrapolate a new picture from it. That's one option which is obviously only suited for one type of artefact -- the movement artefact. And now we all should realize that no proper artefact amplification exists or is at least very hard to implement. How we perceive things needs to be considered as well as what kinds of artefacts are involved.
What's that got to do with audio? Well, replace the term "movement" with "phase shift" and think of high frequency noise (>2 kHz) whose polarity has been inverted (flat 180° shift for all frequencies). The new signal exhibits the exact same signal energy in the exact same frequency/time regions and is very very likely to sound the same to you. Obviously your ear doesn't care about the difference. If you "amplify the difference" you just scale your signal. Scaling it naturally affects perceived loudness. So we do notice a change (in loudness). What does this prove? Difference amplification is meaningless.
Ironically SBR performs quite well in your tests which suggests that your amplification technique does less amplification in the way we're sensible to on those kinds of artefacts than what would be appropriate.
In a nutshell: You assume the percibility of an artefact is doubled by doubling the per-sample difference. This is the strong assumption I meant your tests are based on. I hope you could follow my reasoning above for why difference amplification is a bad idea when it comes to those kinds of audio tests.
SoundExpert tests show that SBR @128 and @192 performs at least as good as other technologies and effectiveness of double-rate SBR increases substantially @320. Do you think it is unusual?