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General - (fb2k) / Re: Gapless Playback
Last post by kode54 -
Video playback is a more complicated matter. You'll need to pre-decode maybe 1-3 seconds of video and audio, and your audio output path will need to report latency to control which video frames to present. When your video loop hits the end of the video frames, it will start over again from the start, filling that frame and audio pool. If the audio is shorter than the number of frames, silence padding will need to be added. Or, you could stretch the audio to match the video duration.

Either way, it's essential that you have a continuous feed of video and audio for your output display, which means buffering. You could save memory by buffering compressed frames, but you'll still need to restart the decoder when you start the video over again. Having 1-3 seconds worth of video already buffered and ready for decoding or already decoded should make restarting the decoder quick enough.

If you use MediaFoundation or DirectShow, or even FFmpeg, do be sure to either implement your own presentation graph, or build a presentation graph that accepts raw, decoded frames, and feed it frames as it asks for them. The key is not to be rebuilding or restarting a presentation graph every repeat, or there will be a significant delay.
Excuse my ignorance if there's an obvious answer to this question, but is there any way to use the metadata mapping in the iPod manager to get an iPod classic to sort by the album artist tag instead of just the artist tag? I was trying to read up on stuff in the features documentation, but I was getting a little confused.
Validated News / Re: WavPack 5.0.0 Final Release
Last post by bryant -
As promised, WavPack 5.1.0 is released. It's available in the same place as always.

Compared to 5.0.0, this is a minor update, but has a few fixes and improvements:

  • added: all new command-line tagging utility (wvtag)
  • added: option to import ID3v2.3 tags from Sony DSF files
  • fixed: fuzz test failures from AFL reported on SourceForge
  • improved: DSD decimation filter (less HF rolloff & CPU use)
  • fixed: non-byte audio depths (12-bit, 20-bit) not showing
  • fixed: rare case of noise-shaping triggering a lossy mute
  • fixed: recognize UTF-8 BOM when reading text files
  • fixed: a few portability issues

Many thanks to all HA members who contributed to this by incorporating WavPack into their projects, with testing and suggestions, and of course by just using WavPack; it is all greatly appreciated!  :)
I have released WavPack 5.1.0 incorporating this command-line program, and have incorporated the ID3v2 import feature into the command-line encoder as well.
Roon is expensive but everyone who uses it pretty much agrees it's the best way to view your music library.  It uses it's own tagging database and pulls data from many places and layers new tags over your tags, thus you can revert back to your own tags at any time.  It integrates with streaming services.  You need one 'server' version to run on the network and other various players can feed from that server.  Many products are now marketed as Roon Ready.  It integrates All Music write ups on artists and bios and discographies and as you browse you can select music to play that you own or from steaming services within that info.  It's been a while since I have read about it so there are likely many new features.  Like I said, it is expensive, but it's almost unanimous that it is the best interface for a music based player.

It does what dbPowerAmp and others do where metadata is concerned, but it does it much better.

You just reminded me of MusicBrainz so I will have to go back for a look.

Well, with 7000 CDs boxed up, hopefully you have a filing system. :^)

General - (fb2k) / Re: Advanced Limiter
Last post by naturfreak -
If the clipping is already in the source file, then you need declipping processing, not simply level limiting.
Limiting of such audio does not restore sample values. The original sample values are lost.

True , but I assumed your original question referred to clean,  unclipped audio, not audio in need of repair.
No, sorry, perhaps I was not clear enough.
For clean, unclipped audio you need a limiter only in case you use additional DSP or preamplifying in the processing chain.
I'm no big fan of limiting as a postprocessing step. Limiting should be done in music production.

I wonder if DAC does some processing steps to conceal clipped samples.
Even expensive Audio-CD-players do not apply additional software-based limiting. How do they process PCM with clipped samples?

The samples shouldn't be clipped, but the peaks of the reconstructed analogue wave form may be clipped if the limit is still 0dB after conversion to analogue, but I assume DACs would have a little headroom.
I think so too.

Foe the record, you can store 32 bit float in a wave file too, It doesn't have to be integer.
I know that, but what is the advantage of that for music listeners? I can only see a benefit in the process of music production.

Additional DSP does alter the level of the audio signal anyway. Does it really matter that much if a limiter decrease the signal level by a very small amount?

I guess from an audio perspective it doesn't matter if a limiter reduces the volume by a tiny amount (I assume you're referring to the reduction described in the OP?), but from an "expected result" perspective it'd be better if it didn't when it's not limiting. Changing the volume changes the md5 calculated for the encoded audio, which should remain the same if the DSP isn't doing any processing.
That sounds like you use the limiter for converting audio files, not just for music playback.
Wouldn't it be a better solution to remove the advanced limiter from converter DSP chain?
foobar2000 has a seperate DSP setup for the converter.
If you don't want any change of your files remove all DSP from converter setup.
Where is the point using a DSP if you don't need it?
I want to have two spectrum elements mirroring each other as a part of my layout but I cant figure out how to do this. Is it possible to do or would I need a plugin?
General - (fb2k) / Re: Advanced Limiter
Last post by naturfreak -
PCM contains no way to distinguish whether a full-scale sample is clipped.  From the perspective of PCM, clipped samples don't exist.
For real audio signals there should no contingous samples at max. or min. value, only single ones.
The original recommendation for mastering audio CD was that the PCM values should not reach limits at any time.

The only thing that we can objectively control is how samples are converted to a waveform:

High quality DAC components should have enough headroom based on how they choose to reconstruct the waveform from the samples.  There's nothing inherently wrong with full-scale samples.
Ok, then. Is a additional limiter as a postprocessing step needed or can we rely on the DAC to handle that?
Audio Hardware / Re: Need help choosing my new headphone!!@
Last post by Hotsoup -
I have the Senn 598 but I don't use them often enough to be confident giving an opinion on their sound. I spent a weekend with them a couple months back and remember thinking the sound was good enough though, and they were very comfortable... for what it's worth.
If you don't want embedded cue-files. just remove them.
Please enable "edit cuesheet" in File>Preferences>Context Menu>Utilities>Edit cuesheet.
Or hold shift and select Utilities>Edit cuesheet and then de-select "enable embedded cue sheet on this file"