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I am not sure if anybody will be interested but those audiophiles who are keen on the best ogg files might be interested in the small modifications to the stock 13.37 slackbuild and large patch for libvorbis 1.3.2 and aoTuV beta 6.03 that I have placed here:
Obviously this replaces the slackware libvorbus package so just be aware of that! Works well against the git FFmpeg and vlc and I have had no problems with other media applications although I would be very interested to hear of any issues.
Upstream vorbis already includes most of the aoTuV changes. aoTuV is better at low bitrate encodes, little to no difference at higher bitrates. http://wiki.xiph.org/Vorbis_Encoders
I am hoping that at least a few people find this useful
I sure did
Even though I did not say so in my first post, I find aoTuV's enhancements great for low bit rate encodings. I tend to use these lower bitrates for my media player. Allows me to stuff 2x the songs on the smallish 1GiB storage.
Though for my personal music collection, I use flac for backup and archival, then oggvorbis at q7. I find it difficult to hear a difference between q7 and flac on most recordings. At q5, I find differences often. Perhaps my low quality speakers, sound card, or ears () aid in the detection.
I am not sure if anybody will be interested but those audiophiles who are keen on the best ogg files might be interested in the small modifications to the stock 13.37 slackbuild and large patch for libvorbis 1.3.2 and aoTuV beta 6.03 that I have placed here:
Obviously this replaces the slackware libvorbus package so just be aware of that! Works well against the git FFmpeg and vlc and I have had no problems with other media applications although I would be very interested to hear of any issues.
Thanks! I'm trying this out right now and will post back if I have issues.
Finally organised myself to redo the patch + slackbuild for Slackware 14 (-current) and libvorbis 1.3.3. Seems to be working well enough on my system, good if somebody else can test it as well?
I suspect that Opus has a slightly different target audio (streaming music, speech as well as general usage) as well as massively less market penetration that Ogg Vorbis. But I would be the first to agree that there are simply too many audio codecs around...
I suspect that Opus has a slightly different target audio (streaming music, speech as well as general usage) as well as massively less market penetration that Ogg Vorbis. But I would be the first to agree that there are simply too many audio codecs around...
Ogg may be still more wide spreaded, but i won't stick to it, as opus is better(for my ears) also for cd-archiving than ogg. Every recent Android-Phone/Tablet has opus-support, all vids on youtube(webm) have an opus-audiotrack ...
Thing that concerns me about Opus is the lack of 44,100Hz (CD-Audio) samplerate support without resampling to 48kHz. Resampling is usually very bad for quality. Whether that matters as much in what is admittedly a lossy format is worth consideration.
I've just built and installed opus/opusfile/opus-tools packages, so I might do some experimenting.
BTW, only 500K of packages, BSD licensed and trivial to build. Is it worth adding them to stock Slackware?
Thing that concerns me about Opus is the lack of 44,100Hz (CD-Audio) samplerate support without resampling to 48kHz. Resampling is usually very bad for quality. Whether that matters as much in what is admittedly a lossy format is worth consideration.
If i get this discussion right, there is no drawback with the samplerate.
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BTW, only 500K of packages, BSD licensed and trivial to build. Is it worth adding them to stock Slackware?
I would like that. To have real benefit from it, progs like audacious/mplayer/xine would need to be compiled against a libopus-enabled ffmpeg. For some reason, ffmpeg is not shipped as separately package.
If i get this discussion right, there is no drawback with the samplerate.
Have a read of the 'sox' manpage, specifically the bit about the 'rate' effect and resampling, note the stuff about pre/post-echo. In short, you can't convert a 44,100Hz sample to 48000Hz without introducing artifacts. But as I said, its a lossy format anyway, so you're going to get artifacts anyway, and for sources that are already at 48K such as DVD resampling isn't going to be needed anyway.
Another interesting consideration which the Xiph guys mention on their faq is that some hardware isn't very good at doing 44,100hz anyway and so you may get better results at your earbuds with 48000.
As for ffmpeg, I suspect Pat doesn't include it because of patent issues. He'd have to disable so much stuff that you wouldn't want to use it in that state anyway. Better to leave people to build it themselves and choose which codecs they want it to support.
P.S. I've only tried listening to opus on some crappy earbuds plugged into an even crappier AC97 based motherboard that picks up all manner of system noises, so its not an ideal platform from which to form an opinion, but what I have heard seemed at least "reasonable", especially at lower bit rates.
In short, you can't convert a 44,100Hz sample to 48000Hz without introducing artifacts. But as I said, its a lossy format anyway, so you're going to get artifacts anyway, and for sources that are already at 48K such as DVD resampling isn't going to be needed anyway.
There is no quality-loss because of resampling in opus, in the xiph mailinglist link above is statet
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Opus files don't have a sampling rate anymore; the internal
representation is most efficiently decoded to 48kHz.
The metadata records the original sampling rate which make it possible
to decode to a file with exactly as many samples as the original
This is not the same as upsampling and downsampling again.
Quote:
Another interesting consideration which the Xiph guys mention on their faq is that some hardware isn't very good at doing 44,100hz anyway and so you may get better results at your earbuds with 48000.
Most soundcards i know are only capable doing 48hz, not 44,100hz. But that's not of interrest for a encoder.
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As for ffmpeg, I suspect Pat doesn't include it because of patent issues. He'd have to disable so much stuff that you wouldn't want to use it in that state anyway. Better to leave people to build it themselves and choose which codecs they want it to support.
ffmpeg without patents-issues/less features is shipped already embedded in Mplayer/Xine.
With the same configure-options, ffmpeg could also be shipped as singel package.
I asked about it here
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