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Recent Posts
4
Movie/Multichannel audio / Re: [REQ] Which codec for (S32_LE) multichannel lossless RECORDING ?
Last post by forart.eu -
According to copilot is possible to install WavPack in OpenWRT using these commands:
Code: [Select]
opkg update
opkg install wavpack

...and, then, is also possible to pipe arecord into WP for realtime compression in this way:
Code: [Select]
arecord --device=hw:0,0 --channels=18 --file-type=raw \
        --format=S32_LE --rate=48000 --buffer-time=20000000 \
| wavpack -i -o "${MNT}/${name}.wv" - 2> >(ts -s >&2) &
recorder=$!

@bryant: is the wp syntax il correct under *nix ?

Last but not least, I've also discovered that TinyALSA is also usable in OpenWRT...
6
CUETools / unable to find the single-track CD with the file browser
Last post by DJ Graco -
I have a few ripped CDs downloaded from the internet. All the rips do not have a corrected offset, even though they were ripped using dBpoweramp, as indicated by the tags. Probably whoever did it did not set the offset for the drive.

They are all single-track CDs.

I would like to apply a fix-offset to these rips using CUETools. However, there is one problem. I have already done similar operations on several multi-track rips and sometimes it worked, sometimes not, depending on whether the disc was in the CueTools database or not. Once I got an error message that the fix-offset could not be applied because the disc was not in the CUETools database and was only in AccurateRIP. I do not know if there are any workarounds for this problem.
Returning to the main issue, the file browser simply does not want to detect these single-track CDs even when I place each FLAC file in a separate folder. I don't know how to correctly indicate the CUETools file for verification or fix-offset operation.
Thanks in advance for any informations.
7
3rd Party Plugins - (fb2k) / Re: External Tags
Last post by Case -
You mention you scan with TPS for 10 seconds. Some streams come with an advert/spoken message when you start the stream. Does this not lead to incorrect gainlevels being detected and written in the tags?
Ideally if the ads play with same loudness as other material, it doesn't. But if the ads are louder it means the ads won't blast your ears when you switch to the station, but the music itself will be quieter.

The partial station scanning is an estimate at best and I wanted to limit the scanning time as 10 seconds already is very long. In that time you have finished scanning an album worth of music.

If the results are bad you can always tweak the gain values manually, as ReplayGain has always intented. Or if you want, you can use Converter or wget or something to capture the station for longer duration, edit the ads away and scan that - then transplant the gain value to station's tag.

Of course TPS writes far too much data in the tags, but easy enough to suppress displaying.
I'd keep component settings as you have configured and instead wipe the extra info away afterwards.

1) it seems that only the option to use a centralized external-tags.db works. How will I be able to send only the processed information (and art) for one station to a friend?
Just few posts up I'm re-explaining to wojak how the default APE tags can also handle that. One tag per station, easy to share if wanted.

2) it would be most convenient if upon creating an external tag for a stream the original %title% (has station name when stream not playing) could be copied to an extra tag %station%.
Tricks like that are outside of the scope of the component. But instead of using the initial "Create External Tags" command, you can use "Edit External Tags" and get normal properties dialog ready to do its thing for all the stations. You can then use 'Tools' -> 'Automatically fill values...' and copy %title% to %station%.
8
3rd Party Plugins - (fb2k) / Re: ReplayGain DSP - Alternative ReplayGain implementation by Case
Last post by Case -
Audio format isn't a factor, loudness depends only on station settings. Last night I saw the DSP get about -5 dB RG value for the station but this morning it consistently fails and reverts to using previous gain, which of course depends on what I was playing earlier.

Having both core ReplayGain and the DSP enabled will cause gain values to be applied twice, so the levels will be quite wrong. It's expected but definitely not the intented way to use ReplayGain. I could force core settings off, or prevent the DSP from working when core settings are not the way they should be. But so far there hasn't been the need.
9
AAC / New Improved AAC Encoder / FAAC — I'm Looking for Feedback
Last post by SYH9999 -
Introduction

Hello everyone, this is SYH. I have been developing a AAC encoder based on FAAC, aiming to improve perceived audio quality and stereo representation. I’d like to share it here on HydrogenAudio and gather your feedback and suggestions. Below are the main features and changes I’ve implemented.

Main Features of the Improved AAC Encoder

1. Consistent 20 kHz bandwidth across all bitrates (default setting into using quality mode)

By default using quality mode, the encoder now preserves frequencies up to around 20 kHz regardless of bitrate.

Users can still fine-tune the bandwidth in the encoder settings if needed.

2. Optimized quantization band “distortion” (leveraging loudness masking)

In loud passages (where sound is already quite intense and tends to mask minor distortions), bits can be allocated more efficiently.

This optimization aims to reduce perceived distortion in the rest of the track, especially in softer or more nuanced sections.

3. Refined PNS (Perceptual Noise Substitution)

Even when PNS is “on,” it doesn’t just stay enabled at all times; it adapts by frequency band.

This approach applies stronger PNS to areas like reverberation and higher frequencies above ~14 kHz, while avoiding unnecessary or incorrect usage in other bands.

4. Additional stereo mode (mixed usage of MS and IS within the same frame)

A new “mixed mode” can utilize MS (Mid-Side) and IS (Intensity Stereo) within the same frame.

By selectively switching per band, the encoder attempts to maintain strong stereo separation while still conserving bits where possible.

5. Enhanced stereo representation

To preserve a more natural stereo field, more of the original stereo information is retained, especially in reverberant or spatially wide sections.

About Bitrate into Quality

-q500 56kbps-80kbps
-q1000 80kbps-112kbps
-q1500 112kbps-144kbps
-q2000 144kbps-160kbps
-q3000 160kbps-224kbps
-q5000 224kbps-

I changed Limit up to -q10000.
I don't recommend if you will use bitrate mode. This encoder may not improved for using bitrate mode.

Expected Benefits and Remaining Challenges

These changes aim to alleviate the common “lack of high-frequency detail” and “weak stereo image” at lower bitrates, while also improving performance at higher bitrates.

Because PNS and stereo mode switching are more complex, there may still be bugs or unexpected behavior in certain edge cases.

Of course YES, I am ready to improve when you take me some issue or bug.

Real-time encoding and lower sample rates (e.g., 32 kHz) have not been exhaustively tested yet, so any related feedback is especially welcome.

What I’d Love Feedback On

Subjective audio impressions
About quantize noise of low level

How does the compare to other AAC encoders (e.g., FDK-AAC, Apple’s AAC)?


About Compatibility and bug reports


Suggestions for further improvement

If you have any ideas on how to refine the encoder, your insights would be extremely valuable.

Download & Test
I’ve prepared a test build (Windows binary)(: (I checked this software is not having malware in Avast!)

The Link is the top!

Conclusion

Thank you for reading this introduction. This project is still in an experimental stage, and I truly appreciate any input you might have—whether it’s about encoding quality, speed, CPU usage, or anything else. I look forward to your feedback and hope we can collaborate to further improve this AAC encoder!

Anyway, Thank you for interested my topic, I hope good impression of this encoder, but Don't Forget Still BETA on this encoder. This is just a AAC-LC Encoder, Please Don't Forget.

 :) And I promise This project will be more exciting than now ! :)

10
Off-Topic / Introduction Post – New Member Joining the Hydrogenaudio Community
Last post by Daisy L. -
Background & Interests:
Hello everyone! :D  I’ve been lurking on Hydrogenaudio for a while, learning about lossless formats, codec comparisons, and audio optimization—finally decided to join the discussion. My interests include:

High-resolution audio (especially FLAC vs. OPUS debates)

Objective listening tests (ABX, blind testing methodologies)

Hardware setups (DACs, headphones, and room correction)

Current Projects:

Ripping my CD collection with EAC (Exact Audio Copy) for archival.

Experimenting with ReplayGain to normalize playback levels.

Questions for the Community:

For classical music, what’s the consensus on 24-bit vs. 16-bit when the source is CD-quality?

Any recommended tools for detecting Ultrasonic Noise in recordings?

Who here attends audio engineering meetups? (Would love recommendations!)