Skip to main content

Recent Posts

1
I could not get tag update working on any Opus combined with Icecast.
First I blamed butt, then LS, but supposedly the problem is elsewhere: https://sourceforge.net/p/butt/bugs/10/

LS has rather poor support on Windows, forget about updating titles from a file.
Still, it's the most powerful broadcasting tool, as it seamlessly integrates Opus and Ogg-FLAC - formats I care about :)
Too bad the tagging issue comes with it...

I'll post the binary if I could finish the Opus 1.2.1 integration into liquidsoap 1.1.1 for native Windows build
2
It seems that you linked volume with quality somehow. I also have a hm1s
https://hydrogenaud.io/index.php/topic,111806.msg922786.html#msg922786

Yes hm1s has low volume, especially when playing high dynamic range stuff like orchestral music and outdoor listening, but still more than adequate for indoor listening and loudness war mixes. I also don't like it only has a 15-step volume control from silence to max. Volume differences for one step is too big that I can't make fine adjustment, which I'd say it has "lower than medium experience".

So I am asking whether redmi note 5 has similar problems. I think that redmi note 5 is perfectly suitable to me and it comes at lowest price and it lasts for life time if not atleast for 10 years. As I have not used redmi note 5 I am requesting help from you.

If you can't give help on redmi note 5 please try to provide help on redmi note 4 because I think that many of you has obtained redmi note 4.
3
General Audio / Re: BBC Radio 3 lossless stream trial
Last post by ev13wt -
Lossless has its purposes as archiving an audio.  And I use FLAC too for my music at home.
But streaming lossless just because “we can”? Sure we can but for what?

Raise a hand who has an issue let’s say with Spotify Premium Vorbis 320 kbps.
I haven’t heard one single issue with it and I’m more than familiar with artifacts of lossy encoders.
If there would be any public test for lossy encoders at ~256-320 kbps (forget about it at that high bitrate) then AAC/Vorbis/Musepack/Opus would have scores like 4.9999… +/- 0.00…01 literally.

Someone would argue that an internet bandwidth is high today.
Well, 50% of all internet connections are still ADSL/VDSL with unstable bandwidth. The average global internet speed is ~7.5 Mbps (as of H1 2017) while there are plenty of DSL connections as low as 3-5 Mbps. Imagine sharing 3-5 Mbps in one family (Netflix, YouTube, Spotify, web surfing, homeworking etc…). 
So, again. Do we really need lossless for streaming if codecs like Vorbis 320k is transparent for 99,9…% of people?

There's an article on medium that Mozilla just posted in their newsletter about this....

https://medium.com/mozilla-tech/mozilla-and-the-bbc-team-up-to-deliver-concert-quality-audio-from-royal-albert-hall-over-the-web-52366d38ab5

Although I believe it's more about a future broadcast :)
From the article:
Quote
The AAC codec is classified as lossy meaning the compression discards some parts of the file that you are least likely to be able to hear. Those losses, however, can diminish the richness and fullness of sound, particularly for serious audiophiles and listeners with well-trained ears.
"Richness", "fulness" and "serious audiophiles"?

What kind of sick joke this article is?


First off, thank you for the link, OP. Very nice!


The way Mozilla describes stuff only stems from not enough education mixed with what "normal" consumers want to hear.
4
IF you eq out a bit of the "Beyer" peak, the Beyerdynamic DT 770 250Ohm is really very comfy for long (8h+) listening sessions.
5
The ABX component doesn't crash. You get the error from Windows as the maximum sample rate supported by DirectSound is 200 kHz. When you play tracks like this outside ABX session foobar2000 automatically enables a resampler to allow playback without errors.

Your workaround suggestion of resampling all files to 192 kHz is valid. You could also keep the files as they are and add a resampler to your foobar2000 playback chain and tick the "Use DSP" checkbox in the ABX component.
6
It seems that you linked volume with quality somehow. I also have a hm1s
https://hydrogenaud.io/index.php/topic,111806.msg922786.html#msg922786

Yes hm1s has low volume, especially when playing high dynamic range stuff like orchestral music and outdoor listening, but still more than adequate for indoor listening and loudness war mixes. I also don't like it only has a 15-step volume control from silence to max. Volume differences for one step is too big that I can't make fine adjustment, which I'd say it has "lower than medium experience".
7
Support - (fb2k) / Re: LAME input bitdepth
Last post by yetanotherid -
I never use Opus myself. Only if I include it when testing as I did earlier, yet almost every time I do I discover it's special in another way. I had no idea it doesn't support 44.1 kHz. I understand it probably doesn't matter, and it'd probably be nice if 44.1kHz somehow just went away, but still.... that was a surprise.

I encoded with Opus again for a look, and this time the number of samples reported by foobar2000 was way different to the source, which makes perfect sense. I assume the same thing happened earlier and when I checked it, foobar2000 was displaying something like this:
Duration 4:49.133 (12 750 757 samples)
And I would have blocked out everything except "757" while switching between the encodes in a playlist, and it was probably bad luck the last three numbers for the Opus sample count were only off by one because then I got to be stupid. I assume that's what happened but I can't check as I deleted the samples and I can't remember which track i used as the source.

I don't normally use ffmpeg's AAC encoder so I'm not fussed there, although I did notice the ffmpeg MP3 had 1D3v2.4 tags whereas the LAME MP3 had 1D3v2.3 tags (which is the tag version I have foobar2000 configured to write), so I probably should investigate that phenomenon.
8
What if I downsampled the original DXD file to 192/24, then upsampled the MQA to 192/24 (to avoid the crashing of the ABX plugin with DXD files), and then use the Foobar ABX test. Would this tell us anything?

So odd that pcm and DSD will ABX fine with the plugin and not DXD.
9
So used the following to upsample

Quote
sox -S 2L-087_06_stereo_DXD_FLAC.mqa.flac MQA_upsampled.flac rate -vsL 352800

Note the L by the -vsL, as opposed to the M from the original link.

It still sounds good to me. Damn good. I really can't hear any differences, even in the silent fade outs.  Very good.

I would like to do a proper ABX test, but the ABX Comparator does not like DXD files, even though Foobar will play them; stupid error. If anyone knows a fix I'd like to ABX properly. In any case though, the linear upsampled MQA to DXD sounds awesome and I can't seem to tell the difference from the ORIGINAL DXD file I downloaded.

10
General Audio / Re: Meridian Audio's new... sub-format called MQA.
Last post by Wombat -
So far undecoded MQA sounds like DXD when upsampled to you. This means low bit, standard samplerate done correctly sounds like DXD.
Can't help with that foobar error but try linear standard SoX upsampling please and try again.