have a "constant quality" mode so e.g. mono speech and 5.1 surround fullband music can be encoded with the same settings and be expected to achieve similar quality (quality as measured by e.g. ABC/HR, MUSHRA, etc). Opus devs have in the past talked about this as "fullband stereo equivalent bitrate."
make as much use of lookahead as possible
allow people to give a hint to the classifier via --speech or --music
include other reasonable settings presently only available via --set-ctl, such as MAX_BANDWIDTH_REQUEST. Forcing the encoder to use higher bandwidths than it thinks optimal is a bad idea, but allowing users to select lower bandwidths is not a bad idea, and along with allowing users to ask for LP/SILK modes can save considerable power on constrained decoders e.g. Rockbox.
I can finally move with my whole music library to 64 kbps! THANK YOU!
2. Opus was getting much better with music files, but there was a fundamental flaw: the frequency had to converted from 44.1 kHz to 48.1 kHz.I've just done a little testing with Foobar 1.3.15 & libopus 1.1.5 and I get 44.1 kHz from a 44.1 kHz ordinary music file. Therefore, from what I can see, this problem has been fixed.
I guess my question would be: is Opus the best encoder for music files and not only for streaming & speech?I've traditionally converted my files using LAME -V2 and I was thinking about moving to Opus @ 128 kbps VBR, hoping to keep the quality (and perhaps improving it), as well as saving some space.
Primo, it's still resampling behind the scenes. Encoder writes original sample rate to Opus file and resamples to 48 kHz. Decoder reads that information and resamples back to the original sample rate. So if you encode 44.1 kHz file, you decode 44.1 kHz file.
Opus outperforms ALL CODECS in both music and speech encoding
Quote from: ziemek.z on 21 June, 2017, 01:14:37 PMPrimo, it's still resampling behind the scenes. Encoder writes original sample rate to Opus file and resamples to 48 kHz. Decoder reads that information and resamples back to the original sample rate. So if you encode 44.1 kHz file, you decode 44.1 kHz file.I'm pretty sure that this is false. Opus re-encodes everything to 48kHz and the music player just reads a tag saying that the original music file's sample rate was 44.1kHz, 8kHz or whatever.If your OS or music player is set to always give 44.1kHz output, that's not related to the Opus codec
I was curious about your statement and I decided to open my converted opus file in Audacity: in fact, it says it's sampled at 48 kHz. I don't understand why MediaInfo says it's 44.1 kHz.
Quote from: Vivadavid on 21 June, 2017, 06:04:28 PMI was curious about your statement and I decided to open my converted opus file in Audacity: in fact, it says it's sampled at 48 kHz. I don't understand why MediaInfo says it's 44.1 kHz.Apparently, the original sample rate is stored in the Opus file's header, and MediaInfo uses that instead of the correct "48kHz" rate.
Quote from: ziemek.z on 21 June, 2017, 01:14:37 PMOpus outperforms ALL CODECS in both music and speech encodingIME, at very low bitrates (under 32kbps) AAC is still clearly better for music, especially when you go down to like 16kbps.For speech it's closer, but Opus can go a bit lower with less annoying artifacts.I didn't compare them at 32+ bitrates lately.
Track: Yaksa - "I Hate You" (metalcore)
Just so I don't get called unscientific or something then get banned from a forum I really love, here's the ABX result XD
Quote from: OrthographicCube on 22 June, 2017, 06:30:01 AMTrack: Yaksa - "I Hate You" (metalcore)I just tried with this track (off Youtube) and I agree with you, Opus sounds better.But then with some "cleaner" sounding music like classical piano music or some pop music, AAC wins, Opus distorts too much.BTW, I'm using FDK AAC in Foobar, because it's the only one that actually encodes to 16kbps for me (12kbps is the lowest, but it botches the beginning of a track).How did you get apple/qaac to encode to 16kbps?