FLAC / bit rate confusion - some files don't play!

taxbuster

New Member
Nov 26, 2011
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I have a new Naim Superuniti which has provided me with a strange problem. I have a variety of high-res FLAC recordings with one different characteristic - bit rate is either 13,824 kbps or 4,608 kbps (per dbPoweramp). Other settings are identical: 24/96, 6 channels, same FLAC encoder, 45% compression. The Superuniti plays the 4,608 kbps files, but not the 13,824s. (Both sets of files play fine in Winamp and Media Center 16.)

I've tried converting them to other formats supported by the SU, keeping the lossless settings the same, but they fail to play (WAV, uncompressed FLAC). AIFF files (at 13,824) play but are missing certain segments of the recording, sounding hollow and missing track portions (that was weird!). Only converting them to a lower res WAV worked.

I cannot find a way of reducing the bit rate on dbPoweramp, but, to tell the truth, I'm now a little confused as to the relationship between the sample rate, sample size and bit rate.
But mostly I'm confused as to why the 13,824s would show up as an 'unsupported format' on the SU and what to do about it and how to reduce them!

Can anyone assist?
Much appreciated!
 

amirm

Banned
Apr 2, 2010
15,813
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Seattle, WA
The "bit rate" is an uninteresting, random value for lossless files like Flac. Because compression is lossless, the encoder is free to pick any bit rate it likes, up to including the uncompressed samples as it sees fit.

That said, the factors that raise the effective bit rate are sample rate, bit depth, number of channels, and difficulty of material. What was the source of the material used for Flac compression? Or did you receive the file that way? If so, it is possible that it uses too value for the aforementioned factors. The only way to reduce that number is to decompress the file, convert it to lower resolution PCM, and then re-encode into PCM.

I am not a user of dbpowermp. But I think I heard that it can upsample the files as it rips them. If so, simply tell it to not do that and rip at 44.1Khz. That should shrink the bit rate.
 

taxbuster

New Member
Nov 26, 2011
4
0
0
The "bit rate" is an uninteresting, random value for lossless files like Flac. Because compression is lossless, the encoder is free to pick any bit rate it likes, up to including the uncompressed samples as it sees fit.

That said, the factors that raise the effective bit rate are sample rate, bit depth, number of channels, and difficulty of material. What was the source of the material used for Flac compression? Or did you receive the file that way? If so, it is possible that it uses too value for the aforementioned factors. The only way to reduce that number is to decompress the file, convert it to lower resolution PCM, and then re-encode into PCM.

I am not a user of dbpowermp. But I think I heard that it can upsample the files as it rips them. If so, simply tell it to not do that and rip at 44.1Khz. That should shrink the bit rate.

Thanks Amir - that makes some sense to me. But what is odd is that it plays in Winamp and Media Center (and Media Monkey now), but not on the (rather expensive) Naim hardware! I'll see if I can manage what you've suggested, though I am a total neophyte at this part of audio.
 

taxbuster

New Member
Nov 26, 2011
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Found the answer, courtesy of the folks at dbPoweramp -- the original FLAC files were recorded for 6 channels (5.1 surround). Reducing the channels to 2 (stereo) did the trick.

The only minor issue is that I had to convert twice: once to a WAV then back to FLAC.
 

amirm

Banned
Apr 2, 2010
15,813
38
0
Seattle, WA
That would certainly explain it. Glad you figured it out. Answering your other question, it is often the case that hardware players are more limited than software. The latter has no restriction of code size and development time so you often find that PC media players will play far more formats and more resilient to errors and special cases than hardware.
 

Vincent Kars

WBF Technical Expert: Computer Audio
Jul 1, 2010
860
1
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I'm now a little confused as to the relationship between the sample rate, sample size and bit rate.

The number of bits is the precision, 16 allows you a dynamic range of 96 dB, 24 = 144 dB.
Conceptually: 24 allows you to hear very tiny details until they sink into the noise floor of your gear.
http://thewelltemperedcomputer.com/KB/Bit1624.htm

Sample rate is about Shannon-Nyquist.
If you sample at 44 kHz, the highest possible frequency you can record is the half so 22
Sample at 88 and the Nyquist is 44, etc.
Higher sample rates allow you to record higher frequencies.
http://thewelltemperedcomputer.com/KB/Sampling.htm

Bit rate is simply the amount of bits per second read by the media player.
Playing CD quality is playing 2 channels with 16 bit word length and 44100 samples per second or 2*16*44100=1411200 bit/s or 1411 kbs.

If you play FLAC you see most of time a value between 700 – 800 as FLAC compresses 40-50%.
This is the bitrate of the compressed file.
However it is lossless so when expanded to raw PCM you have the full unaltered original 1411 kbits.
http://thewelltemperedcomputer.com/KB/BitRate.htm
 

taxbuster

New Member
Nov 26, 2011
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Thanks Vincent - that clears up a lot of confusion in my mind, and also explains why those files I started with would have such a high bit rate --- they have to feed 6 channels at a time.

Now my expectation of that is the same info is being provided to 6 channels as to 2, but some of it is duplicated, so channel 2 receives some of the same info as channel 5 (say), in order to provide the "surround sound" effect.

Is that a correct understanding (at least in lay terms) of how surround works?
 

amirm

Banned
Apr 2, 2010
15,813
38
0
Seattle, WA
Thanks Vincent - that clears up a lot of confusion in my mind, and also explains why those files I started with would have such a high bit rate --- they have to feed 6 channels at a time.

Now my expectation of that is the same info is being provided to 6 channels as to 2, but some of it is duplicated, so channel 2 receives some of the same info as channel 5 (say), in order to provide the "surround sound" effect.

Is that a correct understanding (at least in lay terms) of how surround works?
What was the source of your FLAC files, CD?
 

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