720p H264 + AAC video has itag 22. It used to have 192kbps in the past. But in the spring of 2016, they began to use 128kbps AAC for itNew videos (2017) use 192kbps AAC on itag 22.
Last post by pmattke93 -
I don't know if I'm searching on the wrong places...
I have some 48kHz FLACs in my collection, but for compatibility reasons I want to convert them to 44,1kHz - I am thinking about making this conversion final and not keep the original 48kHz since I don't think I would hear any difference.
Still, I would like to know which resampler is theoretically the one with the best performance in foobar:
There is dbPoweramp/SSRC, and there is PPHS which can be selected in the converter setup. PPHS has an ultra mode, I guess this is something like "better accuracy while sacrificing encoding speed". Is that true?
I just want to get the most accurate result possible, I don't care about encoding speed since I don't use the resampler for live listening but only for a one time conversion of my few 48k files.
Which one would you recommend here?
While searching through the web, I can't find anything helpful that encourages my decision. Any tips from the community?
Thanks in advance.
In Acoustic Guitar I can see 125 kbit/s audio, which is the reduced rate. The bitrates are actually 192 and 125 abr, as reported by, for example, ffdshow, which shows the instantanenous and average readings. Some new videos do still use the higher rate, and there doesn't seem to be any clear reason why one should get the higher, and another the lower. For example, 125 kbit/s, 192 kbit/s, 192 kbit/s. I noticed that one video changed from 192 down to 125 today. Maybe there is a mixture of quality on the YouTube network, and it depends on where the video happens to be served from.
I'm pretty sure the compression comment was in regards to dynamics rather than data, BTW; not that the two don't get conflated.
Don't ask me why I'm even putting this silliness back on the table.
Here's an apparently much needed news flash: Digital audio does not necessarily include lossy compression or added loudness.
There ain't no such thing as a format with zero loss, even in theory as long as we limit ourselves to finite numbers. When you get to real world practice, then that there is no such thing as a format with zero loss is even more abundantly clear.
From the 204 dB of Saturn V to whatever is theoretically closest to negative infinity as dictated by quantum mechanics or whatever, that is a helluvalot of bits and surely way more than 32.
I thought we were talking about audio, which is related to sounds that are potentially audible and non-fatal. Rule of thumb, if you have to conflate audio and quantum mechanics to support your claims, your opponent wins.
Which kind of invites to the hi-rez bandwagon.
The hi rez bandwagon starts at anything > 16 bits, last time I looked around in the real contemporary world.
Going from the sound in the wild to a 32-bit digitized signal is in principle a lossy compression operation.
Excluded middle argument, anybody?
BTW am I conversing with one of those who is so ignorant of how things work they think that analog audio has infinite resolution? ;-)
Truth be known, Vinyl lacks dynamic range as compared to any digital format but the ones that are violently bitrate reduced. Way south of Redbook.
I'm just a normal Youtube user and I was able to upload FLAC audio back in 2013: https://www.youtube.com/watch?v=c3OOgcqm0YkOn the other hand, there are 192kbps AAC.
About 128/192 kbps in 720p video:
Every Youtube quality is associated there with internal integer parameter called 'itag'. Streams with the same itag have generally the same quality (though there can be fluctuations in both directions *). 720p H264 + AAC video has itag 22. It used to have 192kbps in the past. But in the spring of 2016, they began to use 128kbps AAC for it. Some old videos were changed to 128kbps as well, but some not (like in the above example). And programs/plugins, that rely only on itag number, show either 192 or 128 kbps for all such videos.
* - (about 'fluctuations') for example, although Vorbis itag 171 is usually 128kbps, it can be even 176 kbps. Or that 160 kbps Opus can be higher (205 kbps) or lower (102 kbps). Though that could be eventually 'fixed' - an other 207kbps Opus example that I had now has 128kpbs Opus.
Although you can upload flac (and pcm) in hi-res, you don't get hi-res when playing the video, and the original upload is not available to anybody else but google.To Google and you. It should be possible to get original video back using Google Takeout (never tried that though).
To others there is only transcoded quality. Not necessary due to copyright concerns. They are really obsessed by bandwidth reduction, to deliver "hd" quality to more people.
And more info to this quality party. Underrated area imo. With spatial audio there came 4.0 (four channels) 192 - 256 kbps (again, both use the same itag) 48000Hz Vorbis. And Opus (of the same quality). And 5.1 AAC 48000Hz 256kbps.
Aaand even before spatial audio, in the distant autumn 2015, there started to appear 5.1 AAC 48000Hz 384 and 192kbps.
Yes. I removed foo_input_amr and fb2k still can play older amr files
amr support is already included in fb coreNo.
3rd Party Plugins - (fb2k) / Re: ColumnsUI: Is there a way to change font/color on each Filter Column?Last post by musicmusic -
Filters support title formatting, so you can use $rgb() (e.g. something like $rgb(255,0,0,0,0,255)$if2(%genre%,<unknown>) ).
However, you lose proper support for multi-value fields when using title formatting there, unfortunately.
I tried to play older amr files even without foo_input_amr. They can play. That means amr support is already included in fb core.
The problem seems to insist in specific amr files produced by Call Recorder License - ACR. The media info identifies something nonstandard - FileExtension_Invalid
Phone however still can play these amr files.