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General Audio / Re: Fake 24-bit FLAC?
Last post by BrilliantBob -
greynol and dragut, you both have right ;-)

The most common WAV audio format is uncompressed audio in the linear pulse code modulation (LPCM) format. LPCM is also the standard audio coding format for audio CDs.

By the other hand, the WAV format supports compressed audio using, on Microsoft Windows, the Audio Compression Manager.
Development - (fb2k) / Re: ABX module closes on error
Last post by j7n -
Is it possuble set the ABX to release the output device once it is stopped or paused?
No, closing the device would add a distracting delay with CPU load on some ASIO drivers. There is no point keeping the ABX session open without playing for long periods of time.

I can play other files to WaveOut or to another ASIO channel as long as the sampling rate matches.
Audio Hardware / Re: Help me build my first serious music (stereo) system
Last post by ajinfla -
Speaking of speakers...     The biggest difference in "sound character" between different speakers is frequency response, and you can adjust that (to an extent) with equalization.
Yes, the frequency responses aka polar response, the FRs in every direction, are the most correlated to SQ. We hear a combination of direct (on axis) and indirect (off axis/reflections).
EQ is a sound power adjustment. Whenever you cut or boost, both the on and off axis is affected. Beware, especially if based off single point pressure measurement.

There are limits to what EQ can do, especially at low frequencies where the laws of physics come into play.    i.e.  When you boost the bass, you can end-up overdriving the speaker or amplifier.   And, there are room acoustics issues that can't be corrected with EQ.
The key is not trying to fill any nulls by boosting. Cutting peaks, judiciously, is fine. Boosting the overall bass a bit for lower volume listening is fine. As a matter of fact, the Yamaha I mentioned earlier has a clever variable loudness control for just that. Also, boosting the low end of a tower to extend it a bit at low to medium volume is fine. You could counter by cutting say <20hz to keep  excursion in check, especially with vented systems which most home speakers are.


General - (fb2k) / Re: [Request] View ReplayGain Track Gain True Peak Levels in Playlist as dB
Last post by Porcus -
If I choose to prevent clipping according to peak, how much is it reduced then? So much that the maximizing sample hits full scale precisely, or is it kept a bit lower due to safeguard against some Microsoft behaviour near full peak?

This new field gave some interesting info:

I think this should return what is limited by peak in order to prevent clipping, right?
Code: [Select]
%replaygain_album_gain% PRESENT AND %replaygain_album_peak% PRESENT AND "$add($replace(%replaygain_album_gain%,.,,dB,),$replace(%replaygain_album_peak_db%,.,,dB,))" GREATER -1
... assuming album mode is used, then.

Throwing in "%__encoding% HAS lossless AND", returns two thirds of my classical music.

How many tracks hit the ceiling? Replace "album" by "track" returns a third. Again, of my (lossless-encoded) classical music.

Non-classical? Not so much.
General Audio / Re: Downsampling Vinyl-Rips to CD
Last post by FLX90 -
Okay, thank you for the explanation.

So here I have my to batch-files. For CD to ALAC only, without resampling and for Vinyl 24 Bit to 16 Bit.

16 Bit:
Code: [Select]
FOR %%A IN (*.flac) DO ffmpeg -i "%%A" -map_metadata -1 -acodec alac "%%A.m4a" 2>> ffmpeg.log

24 Bit:
Code: [Select]
FOR %%A IN (*.flac) DO ffmpeg -i "%%A" -map_metadata -1 -af "aresample=44100:resampler=soxr:precision=33:osf=s16:dither_method=shibata" -acodec alac "%%A.m4a" 2>> ffmpeg.log

(-map_metadata -1 because I want to clean tags completly with TuneUp.)

16 Bit Input:
Code: [Select]
Audio: FLAC (framed) 44100Hz stereo 1411kbps [A: flac, 44100 Hz, stereo, s16]
16 Bit Output
Code: [Select]
Audio: ALAC 44100Hz stereo 1059kbps [A: SoundHandler (alac, 44100 Hz, stereo, s16, 1059 kb/s)]

24 Bit Input:
Code: [Select]
Audio: FLAC (framed) 96000Hz stereo 4608kbps [A: flac, 96000 Hz, stereo, s24]
24 Bit Output
Code: [Select]
Audio: ALAC 44100Hz stereo 811kbps [A: SoundHandler (alac, 44100 Hz, stereo, s16, 811 kb/s)]

Why are the 16 Bit Outputs always much higher bitrate than the 24 Bit Outputs?

From the log-file:
What does the 128 kb/s mean?
Code: [Select]
Stream #0:0: Audio: alac (alac / 0x63616C61), 44100 Hz, stereo, s16p, 128 kb/s

And what is that overhead?
Code: [Select]
overhead: 0.044341%

I will test the files for clicks.
Hopefully there are none.

Thank you for your help.
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