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1
Validated News / Re: Rockbox 3.14 released
Last post by saratoga -
Such a shame I can't give it a try, as my Fuse's turned into a lil' brick and refused to display any sign of life, for no apparent reason, a few years back!

Sorry for the little digression, but I wonder if anyone else ran into this, with the same player...

Sometimes you can get them to reboot if you plug them back into power for a while after letting it sit.  They're not the most robust players though. I think the flash memory on them tends to wear out over time. 
2
Thank you for the advices!
For now my preference goes to the Sony MDR ZX770, but i'd like to know if the Sennheiser CX 1.00 (29,90€ on Amazon) are good enough so that using loseless formats makes sense. I'm considering used-but-in-good-condition headphones available at Amazon Warehouse deals as well as in other local and 2.0 online shops. 
3
Opus / Re: Opusenc's built-in resampler
Last post by IgorC -
What I see is that this whole topic would have been avoided if OPUS simply supported 44.100Hz audio without the need of resampling.
What I see is that this whole your post would have been avoided  if You just take  5 minutes to read the topic and understand that an error introduced by modern resamplers is much less than quantization  error of lossless material (for both 16 and 24 bits .wav)
It's 100% inaudible.


It's possibly the last thing that holds me back making the switch from AAC to OPUS.
Again. Modern resamplers are transparent today.  It's 2017. Not 1987.


4
Here's a link to the component if you still need it.
esplaylist
5
Opus / Re: Opusenc's built-in resampler
Last post by saratoga -
Opus uses fixed power of 2 sampling rates.  That is not possible to change without making a new codec, officially or otherwise.  So no, it is not going to change.

Yeah right,

You're not understanding me.  The Celt layer in Opus only operates at 48 KHz.  24 KHz can be implemented by decimation.  This is not a matter of my opinion.  It is a simple fact that you can verify by checking the spec. 

I have seen similar stuff being said about countless situations of "this is not going to happen because..." and sooner or later, most often after some backlash, it did happen.

This is wrong thinking.  People say they want things all the time.  To win the lottery, to be young again, etc.  I can assure you that this just isn't going to happen.

This is foolish.  The point of a codec is produce high quality audio, not avoid the need for you to worry about imaginary problems. 

Foolish is to be passive agressive when you can't do constructive criticism.

Nothing I have said is passive aggressive, and you have not presented any constructive criticism.  I am addressing the underlying point clearly and directly.  Fixed sampling rates have a very minor performance hit, but greatly simplies transform coding and reduce latency.  This is a lot to give up just because you don't want to have to do "research" or because you have unfounded "doubts".

Even more, there is no forum TOS that will restrict people posting similar stuff because it goes against the whole nature of forums.

FWIW, you are violating TOS#8 and should probably stop, but I don't see the need to bring that up because I believe you probably didn't mean to.
6
Opus / Re: Opusenc's built-in resampler
Last post by Klimis -
Opus uses fixed power of 2 sampling rates.  That is not possible to change without making a new codec, officially or otherwise.  So no, it is not going to change.

Yeah right, I have seen similar stuff being said about countless situations of "this is not going to happen because..." and sooner or later, most often after some backlash, it did happen. Just, never say never. You have no idea when a person will pop up with a crazy idea.

This is foolish.  The point of a codec is produce high quality audio, not avoid the need for you to worry about imaginary problems. 

Foolish is to be passive agressive when you can't do constructive criticism. This is not going to be the last time you see a comment like mine or a topic like this for very good reasons. Even more, there is no forum TOS that will restrict people posting similar stuff because it goes against the whole nature of forums. You just cannot call somebody/everybody's claims and opinion foolish. Also, imaginary is the elitisism that alot of members around the forum feel that they are entitled to, yet it's unflattering for the image of the forum at best (to be worded in a way that is respectfull to all members of the forum).
7
This is a great site for comparing and viewing the various higher quality headphones of all types.

I personally prefer the Shure headphones for both in ear and over the ear, but they are a little on the pricy side.

https://www.headphone.com/
JXL
8
General A/V / Re: Swarm Recording
Last post by saratoga -
AFAIK if the cell phones are all on the cell system, or running off of a common WiFi Hotspot, they are automatically time synchronized with each other. Precisely!

While you could make a phone that worked like this, the A/D+D/A clocks are not going to be derived from the remote Wifi or 4G signal in a typical phone.  Instead they'll be derived from the same oscillator as the CPU.  Usually you don't actually have access to the underlying RF clocks on a SOC, so time aligning audio will be complex (but of course entirely possible in post processing). 

FWIW, I have wondered if anyone would consider making a SOC that worked like you're proposing for stuff like a Chromecast where you want to do multiroom audio.  You're right that it should be possible. 
9
If the tracks you converted are, in the playlist, the only ones with mp3 extension, try to save the playtlist (in fpl), open it as text with a good notepad (like Notepad++), use the function for replace all .mp3 strings with .wma strings, and save the file.
If the tracks you converted aren't, in the playlist, the only ones with mp3 extension, copy the interested entries (the tracks you converted) of the playlist in another empty playlist, save it, and use the method I suggested to replace the strings.
Now you can load the modified playlist, with extension replaced, in the original playlist to replace the tracks with playback error due the changed extension of the files.

A not so completely automatic method, but very simple and fast to replace in few time also a large amount of entries of a playlist without recreate the structure of the involved entries.
This method supposes that, in the paths of the files, the string .mp3 is only in the extension (so neither in the folders names nor in the filenames).
10
Opus / Re: Opusenc's built-in resampler
Last post by saratoga -
What I see is that this whole topic would have been avoided if OPUS simply supported 44.100Hz audio without the need of resampling.

Why stop there?  If Opus was a lossless codec then people couldn't complain about quantization error at all!

I feel like it's a matter of time until it eventually gets supported (officially or not).

Opus uses fixed power of 2 sampling rates.  That is not possible to change without making a new codec, officially or otherwise.  So no, it is not going to change. 

The less layers of change applied to the source the better outcome to the final product (audible or not), with less doubts and need for research.

This is foolish.  The point of a codec is produce high quality audio, not avoid the need for you to worry about imaginary problems.