Codecs and settings for 64kbit/s SE listening test, criticism required Options

[Serge Smirnoff]

As SoundExpert now has pretty stable flow of volunteer testers it is time to update codecs in all bitrate sections. After a short discussion at Russian bit-torrent tracker we decided to update 64-96-128-192-256-320 sections first. For the 64kbit/s section five codecs were chosen:

Fhg AAC (2012-06-24) - 59.2 kbit/s (fhgaacenc --vbr 2 se_ref.wav)
QAAC TVBR (v1.42) - 59.6 kbit/s (qaac --he -v56 se_ref.wav)
Nero AAC (v1.5.4.0) - 60.1 kbit/s (neroAacEnc.exe -q 0.25 -if se_ref.wav -of out.mp4)
Vorbis (Xiph 1.3.3) - 60.3 kbit/s (oggenc2.exe -q-0.3 se_ref.wav)
Opus (libopus 1.0.1) - 59.9 kbit/s (opusenc --bitrate 59 se_ref48.wav out.opus)
Conversion chain for Opus: 44.1/16 -->> 48/24(Audition CS6) -->> opusenc -->> foobar2000(48/24) -->> 44.1/16(Audition CS6)

Bitrates are calculated on the basis of nine SE test samples concatenated. Fortunately first two codecs have close values of resulting bitrates at their corresponding discrete quality settings. Other contenders were adjusted to have close values.

Did we miss something?

[lvqcl]

It's probably a very daunting task to get good results going beyond 128 kbps. Have you performed successful listening tests at high bitrates in the past?

Ratings above 5-th grade mean that the devices/technologies have some quality headroom, their artifacts are beyond threshold of human audibility. Testing files of such devices are processed additionally - sound artifacts are amplified to the extent when they could be heard by ordinary listeners.

(from soundexpert.org, emphasis mine)

This post has been edited by lvqcl: Oct 14 2012, 22:48

[Kohlrabi]

It's probably a very daunting task to get good results going beyond 128 kbps. Have you performed successful listening tests at high bitrates in the past?

Ratings above 5-th grade mean that the devices/technologies have some quality headroom, their artifacts are beyond threshold of human audibility. Testing files of such devices are processed additionally - sound artifacts are amplified to the extent when they could be heard by ordinary listeners.

(from soundexpert.org, emphasis mine)

How do you algorithmically selectively amplify artifacts, other than modifying the encoders to work worse than they'd normally would?