For context, LDAC is one of the few wireless audio codecs stamped Hi-Res by the Japan Audio Society and its encoder is open source since Android 8, so you can see just how long Windows is sleeping on this. I’m excited about the incoming next gen called LC3plus, my next pair is definitely gonna have that.
For context, LDAC is one of the few wireless audio codecs stamped Hi-Res by the Japan Audio Society and its encoder is open source since Android 8
LDAC is great, but simply stating that the encoder is “open source” is quite misleading (while technically correct). The codec is owned by Sony and heavily licensed. It’s a savvy business move of Sony to make the encoder free to use though, so everyone else can support their standard while charging manufacturers who want to integrate it into their headphones.
If we want a really free and open high quality codec, we should push for opus support via bluetooth
Yes… I made double sure to mention ‘encoder’ between that.
Xiph really won the lossy codec scene with Opus and I transcoded all my junk to that format. Hitting (my personal) transparency on 128k vbr is flat out impressive and it warms my heart that corpos won’t have a reason to collect taxes for basic things like audio codec. However it’s a different story with bluetooth audio codec in which I hope will change.
Xiph really won the lossy codec scene with Opus and I transcoded all my junk to that format. Hitting (my personal) transparency on 128k vbr is flat out impressive
Same here. I’ve left myself a bit of a safety margin at 144k vbr, but having my whole library at transparent quality AND portable size is very convenient.
Though, now that opus 1.4 is out I feel a bit of anxiety whether i should re-encode everything from flac->opus1.4
Which tool do you use to re-encode everything to opus ?
I tried with ffmpeg and it works but I had many issues with covers.
same as @denissimo@feddit.de I use foobar2000 + wine. ffmpeg is alright, but fb2k is very convenient (especially for replaygain tagging). Afterwards I usually give the files a Picard treatment to get proper tags + covers.
I use foobar2000 + wine
Check out Strawberry it’s essentially the Linux native version of foobar2000.
does it support foobar2000 plugins?
probably not, since those are windows dlls. So here’s a short list of what I’d want from a fb2k replacement:
- a UI plugin with the power and flexibility of Facets/Refacets
- browse library by folder structure OR tags (most only do one or the other)
- powerful query language to actually find what I’m looking for
- binaural stereo for headphones plugin
- convolver
- convert to opus and replaygain scanning
- DR Meter
- handle my >100k tracks library without constantly crashing or being incredibly slow
Most alternatives I’ve tried can’t even deliver on half of those.
I use foobar2k via wine. Yes, you may stone me. Tip: You will save heaps of space by not embedding the cover on each file, just put a cover.jpg in the albums folder, virtually any player will pick it up.
Tip: You will save heaps of space by not embedding the cover on each file, just put a cover.jpg in the albums folder, virtually any player will pick it up.
Except when streaming the file or copying a random file to another location. embedded art is pretty convenient, 500x500 is plenty large enough and doesn’t take a lot of space (~50KB)
Use opusenc directly. It preserves covers and the CLI is literally
opusenc --bitrate B INPUT OUTPUT.
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Transcoding to a (for them) transparent lossy result is perfectly fine if all you do is listen. I couldn’t care less about “audio qualities” that I cannot hear.
If we want a really free and open high quality codec, we should push for opus support via bluetooth
Isn’t the new default codec in BLE Audio LC3 free and open and high quality? And it’s required for BLE Audio support, so there will be more and more devices that support it.
LC3plus isn’t really HiFi. It’s designed to be low-complexity & low energy: https://hydrogenaud.io/index.php/topic,122575.0.html
I recommend you all to switch to Pipewire. Most bluethooth problems are fixed.
Most destros default to Pipewire, even Debian 12.
Isn’t that standard on most linux systems?
if your distro is not using Pipewire youre using a shitty one
I’m so confused, please don’t confused a new linux user it doesn’t help me
You can find out if you have pipewire or pulseaudio by using
pactl info | grep "Server Name"You lost me
I believe @twei meant, if you open a terminal and put in the following command:
pactl info | grep “Server Name” the output should tell you what sound server you are using.Or an even better and less nerdy was is just using. System taskmaster and seeing pipewire
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Where the hell do you still buy Anker headphones? I’m in germany and holding on to my soundbuds flow because nothing else even comes close to comparison. I get Anker chargers and adapters and cables and whatever but nothing audio related to a point I thought they gave up on the whole branch.
They use the Soundcore branding currently for their audio products but it‘s the same company
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Just checked. They are and are all called “Soundcore by Anker so and so”
They are even shipping through Amazon even if bought through Anker/Soundcore website.
Micro center carries soundcore products. Where I bought my current P2-L headphones, couple years back.
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Sony did drop the ball with LDAC quite quickly, it could’ve been the new standard.
But with the release of the WH-1000XM3s (or was it the 4s?) they basically made most of the selling points incompatible with LDAC, so now almost no one uses it anymore.
Yes, LDAC and multipoint do not mix hence I’m looking forward to LC3plus that replaces it. To be fair it’s not a big issue to roll back to AAC or even SBC to use multipoint, because you probably aren’t gonna notice a difference when you don’t listen to high res apps like Tidal. It also should be known that a good codec does not fix mediocre drivers and/or chips. Regardless, Linux shines in letting you use a feature you did pony up for. :)
AAC hurts my ears. Not sure why since I can’t hear a difference between it and LDAC without listening very carefully, but after half an hour or so I need to switch it to something else because it becomes more and more uncomfortable.
Switching between LDAC/multipoint mode means rebooting the headphones and connecting them again, so it’s a massive hassle. That makes multipoint absolutely useless to me. I personally won’t be buying sony headphones (or anything else that comes with an app) in the future because of that.
Aac has a higher frequency response and I think some decoders don’t filter transients as well with a fir filter. I’ve noticed this too.
I actually prefer AptX HD but I wish my Android would default to it instead of LDAC
For me it all depends on what I connect to, my head unit in the car defaults to AAC, my portable BT speaker is using LDAC, my bookshelf is using AptX-HD.
Thanks Android for supporting all of them automatically!
I’m still wondering how to make my headphones work on pop os without crackling
You will not believe this. Solution was adding a line in some config file.
Are you thinking of the standby timeout? Cause I get static on my speakers on any and all distros when no audio is playing. It always happens after 5 seconds of silence. Kinda infuriating that I have to do this on EVERY SINGLE DAMN INSTALL.
For Pulseaudio:
Quickfix (until reboot):sudo su echo 0 > /sys/module/snd_hda_intel/parameters/power_savepermanent fix is to add the line:
options snd-hda-intel power_save=0
to the file /etc/modprobe.d/alsa-base.confFor pipewire:
create folder /etc/wireplumber/main.lua.d/ if it does not exist
if you had to create it yourself just copy over the file from /usr/share/wireplumber/main.lua.d/50-alsa-config.lua
otherwise it probably is there already then just edit it
pretty much at the bottom there is a line that says “session.suspend-timout-seconds”
uncomment it and set its value to 0
then rebootPretty sure this used to be the fix for me:
https://forums.linuxmint.com/viewtopic.php?t=314918
( scroll down to comment about default.pa and tsched=0 )
But I just checked my default.pa and it is stock values, so I am not sure anymore
When experimenting, setting cpu governor to performance also helped.
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Used to have that problem on suse, it was something to do with pulse but I can’t recall, solution was in ArchWiki
Any way to see which bitrate is currently being used? I know you can set it to use only 909kbps, 606kbps or 303kbps in the wireplumber config, but I am curious which bitrate the adaptive mode (usually) uses.
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I definitely love it for the “it just works” (or rather, you have full control to make it work) factor!
I’m not familiar with the latest in BT audio, but isn’t the standard still sub-par in that it has very limited overall frequency bandwidth, resulting in deep sacrifices to fidelity?
I recall a detailed analysis of different BT audio codecs a while back, and the spectrum analysis always showed relatively high noise floor and frequency roll-off (hi-cut/low-cut) within the threshold of human hearing (though admittedly close to the limits). Also, I recall (and this could just be the 2016 tech I am familiar with) that overall bandwidth was limited in that if you played something with low frequency tones, the upper frequencies were dropped, or vice versa. I used to confirm this by using a flat EQ setting, then boosting any range, and you could easily detect the loss of frequency response in the adjacent or distant ranges.
Is this a thing of the past now?
I love it because “it just works” (or rather, you have full control to make it work)
This is such a perfect microcosm of the hilarious irony of Linux fanatics.
Tell me you’ve never actually tested the quality of a codex and how it’s used without saying it.
