I didn't find that SoX do dithering by default.
I am using this one (swr):
ffmpeg -i "input.wav" -af aresample=resampler=swr ^
-acodec alac -ac 2 -ar 44100 -sample_fmt s16p output.alac
It is now apparently impossible/illegal to have multiple levels of popup windows where one is a parent of another.
Right now, the main foobar2000 window is the parent of the properties dialog, and the properties dialog is the parent of the tag update progress dialog. When the progress dialog disappears, the main window is incorrectly moved behind other windows.
The only mitigation I could come up with is to make the progress dialog owned by the main window - but then you can bring the properties dialog in front of the progress dialog which is not what I had in mind.
Let's wait and see if this gets fixed by a Windows update. Certainly there are other apps affected by this, not just foobar2000.
Here's my dilemna, i have a few files that were converted to mp3 160cbr using lame, 3.998 years ago, i then converted the files from mp3 to wave using dbpoweramp 15r recently, wave settings 'wave' uncompressed 16bit(cd) , channels (2 stereo cd), 44,1khz. the results were interesting.
I compared the wave track to it's mp3 equivalent, and straight away i noticed that the wave track sounded louder, and the drums and snares sounded better, like the wave container, was adding a slight reverb and gain to the same track. The track illustrated no compresssion had taken place, [as duly understood].
I looked at the spectral analysis using spek, and the was no change in the waveforms between the mp3 file and wave file, i was looking at the frequency cut-off and amplitude.
I tested both tracks using headphones and speakers, with no eq, at the same volume, and when i clicked on the wave file, it would instantly be louder than the mp3 and sound better.
It may be in the mind, but was wondering if some people who have a similair set up could test and publish results. please don't reply with the rhetoric, which is banded over the internet, it will sound worse, actually do the test and post findings.
I also found that going from lame v0 To wave made virtually no difference to the sound quality and gain, and whatever you do after going from mp3 to wave, do not try going to flac, as the quality does deteriorate.
The last note is i used r&b, slow jams and something more uptempto, old skool funk and soul 80's, rock 'bowie' as samples, also about another 14 more tracks, over the last two weeks, taking breaks during lsitening.
I have some 192 kHz / 24 bit Vinyl-Rips in FLAC and want them to downsample to 44.1 kHz / 16 bit for my iPhone.
I would like to do that with FFmpeg + SoX.
So is that line correct?
ffmpeg -i input.flac -af aresample=resampler=soxr -acodec alac -ar 44100 -sample_fmt s16p output.m4a
I read about dither with downsampling.
But that is done with SoX by default?
IMO lame v3.100 is the best .mp3 encoder.
- aaa.wav before encoding
- out.mp3 after encoding with my above mentioned lame settings.
Also make sure you have audio driver installed for your device as without driver you will only be able to play up to 24/96 resolution.
Then review your foobar DSP settings and remove everything unnecessary.
The rest depends on audio signal source you feed and playback hardware (DAC, AMP, speakers, cables)
If you want to know what lame.exe really can do, you must recompile the source code with all the dev settings unlocked. Now I realized the .mp3 format is not dead; can be used even for high quality encoding, with the .mp3 file size +7x smaller than the .wav format.
By the way, use some good quality speakers to hear the differences; my subwoofer make 20 Hz audible signal and the best vibration output is at 34 Hz -3dB.
I made a small bat file to automate the finding of the best scale when I encode with lame.
@echo offP.S.- Before the .mp3 encoding, I used some wave editor to make sounds better for me.
echo Ý ÚÄÄÄ¿Þ
echo Ý LAME v3.100 64bit unleashed ³ û ³Þ
echo Ý ÀÄÄÄÙÞ
:: highpass filter disabled. polyphase lowpass filter disabled
:: using short blocks if better
:: interchannel masking ratio: 0.0002
:: using joint stereo for better compression
:: for access to dev settings (--help dev) put "#define _ALLOW_INTERNAL_OPTIONS 1" in parse.c and compile
:: default lame psychoacustic tuning: --ns-bass -0.5 --ns-alto -0.25 --ns-treble -0.025 --ns-sfb21 0.5
:: adjusted masking (more clear and selective sounds, noise out): bass=-1 dB, alto=-0.5 dB, treble=-0.275 dB, sfb21=0 dB
lame.exe -mj --ns-bass -0.5 --ns-alto -0.25 --ns-treble -0.25 --ns-sfb21 -0.25 --short --verbose -q0 -b320 --cbr -c --resample 48 --highpass 0.001 --lowpass -1 --clipdetect aaa.wav out.mp3 --bitwidth 24 --interch 0.0002 --scale 1.5
Set /P _link=new scale:
lame.exe -mj --ns-bass -0.5 --ns-alto -0.25 --ns-treble -0.25 --ns-sfb21 -0.25 --short --verbose -q0 -b320 --cbr -c --resample 48 --highpass 0.001 --lowpass -1 --clipdetect aaa.wav out.mp3 --bitwidth 24 --interch 0.0002 --scale %_link%
Set /P _abort=exit? (y, *):
If /i "%_abort%"=="Y" goto step2
If /i "%_abort%"=="y" goto step2