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1
I also compared the three resampling algorithms available in stock foobar:

 * dBPoweramp/SSRC
 * PPHS
 * PPHS + Ultra Mode

Converted a 48k file to 44,1k and then back to 48k, then I compared the results and the original in a DAW, twisted the phase of the original and played both together to see what's left:

For PPHS (with and without Ultra Mode) it was mainly noise at about -100dB (at the loudest parts of my test file) and below. With some more high frequency diff at I think about 18-20kHz, where the delta was going up to -80dB. But still of course. Nothing to fear here :D

For SSRC, I was at first surprised that I still could actually quite good listen to the music in the diff between original and twice resampled: Playing both alternating, there was no audible difference for me, but playing the diff sounded a bit like I would grab an EQ and cut off some bass and low mids. I guess there were just some phases different from the original. As far as I know, resampling algorithms often (or always??) are implemented by applying a low pass filter to a zero-stuffed signal. So phase could of course change here, depending on which low pass filter was used. Just some thoughts of mine, don't know if they are true :D can somebody verify?

I know now that both alternatives offered by foobar seem to be transparent to me. But still I want to make a decision. SSRC or PPHS? It's of course not a topic about audible problems. Much more about myself feeling better if I know that I used the resampling method with the best accuracy.

After my tests described above, I would currently go with PPHS + ultra mode, because the noise resulting from the diff was still a bit lower than with PPHS without ultra mode. But due to the phase changes described above for SSRC, I can't determine its quality and there were places on the web were I remember I have read that SSRC would be better than PPHS (but I think the posts / articles where are read that are quite old)
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I wouldn't worry too much about it...  I've never heard any difference when converting in either direction between 48kHz (for video) and 44.1kHz (for CD), no matter what resampling software I happen to be using.

...And I ONLY resample when I need 48kHz for video, or 44.1kHz for CD.  If I'm making audio files (MP3s, etc.) from video I leave them at 48kHz.  

I've seen graphs that show a difference, but I'm pretty sure any competent DSP programmer can write a resampler that's better than human hearing, as long as you are not downsampling down to the point where you loose high (audio/audible) frequencies.
3
WavPack / Re: How about multi-threaded wvunpack?
Last post by lvqcl -
FFmpeg's wavpack decoder is multithreaded and is much faster than wvunpack even with 1 thread.

I just tested wavpack 5.1.0 and ffmpeg 3.3.1 and ffmpeg was noticeably slower in single-thread mode.
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WavPack / Re: How about multi-threaded wvunpack?
Last post by saratoga -
Hi, is there a reason not to use multiple threads in wvunpack to speed up the unpacking of lossless WavPack files to WAV? If not, could you make a multi-threaded unpacker?

I think most common programs already do this.  Foobar, dbpoweramp, etc. 
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WavPack / Re: How about multi-threaded wvunpack?
Last post by atomnuker -
FFmpeg's wavpack decoder is multithreaded and is much faster than wvunpack even with 1 thread.
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WavPack / Re: How about multi-threaded wvunpack?
Last post by lvqcl -
Too few people use Wavpack to encode 10MHz data, I suppose...  ;)
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WavPack / How about multi-threaded wvunpack?
Last post by Funk -
Hi, is there a reason not to use multiple threads in wvunpack to speed up the unpacking of lossless WavPack files to WAV? If not, could you make a multi-threaded unpacker?
8
General Audio / Re: How to obtain true hi rez audio via YouTube?
Last post by quadH -
720p H264 + AAC video has itag 22. It used to have 192kbps in the past. But in the spring of 2016, they began to use 128kbps AAC for it
New videos (2017) use 192kbps AAC on itag 22.
9
Hi all,

I don't know if I'm searching on the wrong places...

I have some 48kHz FLACs in my collection, but for compatibility reasons I want to convert them to 44,1kHz - I am thinking about making this conversion final and not keep the original 48kHz since I don't think I would hear any difference.

Still, I would like to know which resampler is theoretically the one with the best performance in foobar:

There is dbPoweramp/SSRC, and there is PPHS which can be selected in the converter setup. PPHS has an ultra mode, I guess this is something like "better accuracy while sacrificing encoding speed". Is that true?

I just want to get the most accurate result possible, I don't care about encoding speed since I don't use the resampler for live listening but only for a one time conversion of my few 48k files.

Which one would you recommend here?

While searching through the web, I can't find anything helpful that encourages my decision. Any tips from the community?

Thanks in advance.
10
General Audio / Re: How to obtain true hi rez audio via YouTube?
Last post by j7n -
In Acoustic Guitar I can see 125 kbit/s audio, which is the reduced rate. The bitrates are actually 192 and 125 abr, as reported by, for example, ffdshow, which shows the instantanenous and average readings. Some new videos do still use the higher rate, and there doesn't seem to be any clear reason why one should get the higher, and another the lower. For example, 125 kbit/s, 192 kbit/s, 192 kbit/s. I noticed that one video changed from 192 down to 125 today. Maybe there is a mixture of quality on the YouTube network, and it depends on where the video happens to be served from.