Last post by ThaCrip -
Playing around with my Klipsch Pro-Media speakers on the PC with the 'release demo page' (the OP linked to) I can notice differences on the speech samples (like while the audio file is playing I switch back and fourth between Opus versions at the following rates)...
-9kbps = noticeable difference between v1.2 and v1.3 as it's a bit less muffled/more clear in v1.3. between v1.1 and v1.2 I don't notice any difference on this 9kbps setting.
-12kbps = noticeable difference between v1.1 and v1.2 but nothing between v1.2 and v1.3 as far as I can tell. NOTE: I would say this one is harder to spot a difference between v1.1 and v1.2 vs the 9kbps and 16kbps settings I mentioned which are easier to spot.
-16kbps = noticeable difference between v1.1 and v1.2 (basically less muffled and a bit more clear sounding) but I don't notice anything between v1.2 and v1.3.
-20kbps = things start to get harder to notice between the versions but I think I can spot some difference between v1.1 and v1.2.
-24kbps = I can 'maybe' notice a difference between v1.1 to v1.2.
-32kbps/40kbps = I can notice some change in sound between v1.2 and v1.3. basically a bit more clarity I guess I could say.
-48kbps = I can't say I notice anything here between Opus versions.
NOTE: from what I have noticed so far in general with my limited testing with my speech files... I prefer no lower than 13kbps with Opus v1.3 and 14kbps with Opus v1.2.1 for general speech at the minimum.
NOTE: I know headphones are optimal but I just thought I would list the above info for whatever it's worth
I hope to obtain good results with all filters OFF and with the psy fine tuning (interchannel masking ratio and sticking around with the psy masking values). The default settings output something crappy and the differences are obvious for any real Hi-FI audio system. The increment of the psy masking values is +/-0.25 dB, added/subtracted to the default psy values. The sfb21 is treble dependent and, along with the low pass filter, cuts the high frequency sounds brutally. Working in progress.
High frequency content (>16kHz) is difficult to encode, this is why the low-pass filter removes them. As you say yourself, you cannot hear any difference. Only the spectrogram shows the filter, but having nice spectrograms should not be the aim. (If it is, forget mp3 altogether and just save a png of the original spectrogram )
When you disable the low pass filter, LAME starts wasting bits on encoding the high frequency content (which you cannot hear), and then it has less bits to encode the low and mid frequency information, so it will sound worse.
Thank you for sending me the samples. I cannot ABX any difference between the wav and your mp3, but that is also the case with the LAME default settings. And the default settings preserve the original 44.1kHz sampling rate. Resampling to 48kHz will not improve quality.
Last post by MetaPixel -
I don't know how these equipment of yours works, but if bandwidth isn't a problem the Opus codec is one of the best options as it can go from very high quality transparent audio down to intelligible 6kb/s speech and have very low latency (20ms by default, can go even lower or higher if needed). The crackles you talk about... if there are crackles in this analog transmission I guess there would be some packet loss, Opus can compensate for packet loss as well using forward-error correction. Here are some examples, including one with the mentioned forward error correction. Note that the latest versions sounds even better than these examples.
If the bandwidth ever gets to be a problem, consider Codec 2 as well. It is also open source, designed for low bandwidth HF/VHF digital radio and can do good enough speech with lots of noise even at around 2kb/s.
Last post by kode54 -
I think he was trying to imply it's not lossless because you can stuff any data you want into it, including decoded MP3 files. But in that case, CDs aren't lossless either, since you can burn audio tracks from any lossy source you want.