Tidal chucking MQA?

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#65
acko said:
If digitally mastered at 44.1k 16-bit, like those from yesteryear especially when CDs first came out. Nowadays studios may go for 96k 24-bit or higher fs master and then churn out other formats for distribution
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Only in recent times. Most music people on this forum buy was recorded on tape or older digital systems, often 48/16.
Then many of the analogue masters were lost in fires, so they are working with copies of copies.
 
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#69
Well, it wasn't any "master" before either as a partly lossy coding (MQA) was used to distribute the media. Master was just a chosen name by Tidal, perhaps suggested by the MQA gang...?

With flac you will be sure of that linear PCM is transferred. If the recording was any good or/and if the mastering process in the end of the production chain forked up'd the SQ, you will hear it in its full glory.

Tidal quality tiers now read as: "lossless, HiRes FLAC, and Dolby Atmos" which is of course a bit stupid as I believe all these are lossless, have equal resolution (for humans) but different dynamic headroom... but its hard... 🙂

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#70
There is a difference between FLAC and Studio Master FLAC - see Linnrecords, attached. The latter is what Tidal should have considered as replacement for MQA. The rest like MAX (FLAC) for Tidal and HiRes (FLAC) for Quboz and others are just marketing bs. CD quality maybe good enough if recording and mastering done properly as you have also indicated.

I will ask them. Usually, none or very delayed responses with vague info so will unsubscribe first and then some machine or AI bots (nowadays) will implore reasons quickly 🙂
 

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#73
Lets hope not... let it RIP...

The spec:

https://en.wikipedia.org/wiki/FLAC

Some of its capability:

<4>​
Sample rate:
  • 0000 : get from STREAMINFO metadata block
  • 0001 : 88.2kHz
  • 0010 : 176.4kHz
  • 0011 : 192kHz
  • 0100 : 8kHz
  • 0101 : 16kHz
  • 0110 : 22.05kHz
  • 0111 : 24kHz
  • 1000 : 32kHz
  • 1001 : 44.1kHz
  • 1010 : 48kHz
  • 1011 : 96kHz
  • 1100 : get 8 bit sample rate (in kHz) from end of header
  • 1101 : get 16 bit sample rate (in Hz) from end of header
  • 1110 : get 16 bit sample rate (in tens of Hz) from end of header
  • 1111 : invalid, to prevent sync-fooling string of 1s
<4>​
Channel assignment
  • 0000-0111 : (number of independent channels)-1. Where defined, the channel order follows SMPTE/ITU-R recommendations. The assignments are as follows:
    • 1 channel: mono
    • 2 channels: left, right
    • 3 channels: left, right, center
    • 4 channels: front left, front right, back left, back right
    • 5 channels: front left, front right, front center, back/surround left, back/surround right
    • 6 channels: front left, front right, front center, LFE, back/surround left, back/surround right
    • 7 channels: front left, front right, front center, LFE, back center, side left, side right
    • 8 channels: front left, front right, front center, LFE, back left, back right, side left, side right
  • 1000 : left/side stereo: channel 0 is the left channel, channel 1 is the side(difference) channel
  • 1001 : right/side stereo: channel 0 is the side(difference) channel, channel 1 is the right channel
  • 1010 : mid/side stereo: channel 0 is the mid(average) channel, channel 1 is the side(difference) channel
  • 1011-1111 : reserved
<3>​
Sample size in bits:
  • 000 : get from STREAMINFO metadata block
  • 001 : 8 bits per sample
  • 010 : 12 bits per sample
  • 011 : reserved
  • 100 : 16 bits per sample
  • 101 : 20 bits per sample
  • 110 : 24 bits per sample
  • 111 : 32 bits per sample

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#78
There is a difference between what someone observes on an oscilloscope and what information is lost or corrupted. Quantization affects the original information and not in a good way

Whether 16 bits is enough is debatable. 24 bits is enough IMO. It also depends on how accurate the quantization is in practice. Obviously sample rate and cut-off filtering go hand in hand and this is a big source of practical information loss or corruption.

So IMO it is both theoretically wrong and practically wrong to assert that CD is not lossy.
 
#79
But by that standard all audio formats are lossy. After all, they all have their bandwidth limitations and dynamic range limitations. Tape hiss is a thing. Records are pretty noisy too and even a good cartridge might sport 20-40 dB of channel separation, meaning that 1-10% of the left channel information ends up in the right channel on playback. I suppose you would argue that this isn't a loss. I would argue that it most certainly is corruption of the original signal.

As shown in the video, dithering and noise shaping can push the quantization noise well below audible. Even with 16-bit.

Tom
 
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#80
A few points:

FLAC is a lossless compression system kind of like zip, only more effective for music files.

Regarding dither and noise shaping, quantization noise/distortion may be pushed sufficiently below audible, but what replaces it may still be somewhat audible. There are sophisticated adaptive noise shaped dithers available today for mastering. One or the other algorithm may be used because it sounds better with a particular piece of music.

The other thing is that CDs are still recorded with less headroom for intersample overs and more overall compression (because of the limited bit-depth) as compared to hi-res releases of the same music. The differences are measurable with modern loudness metering (LUFS meters).

That said, well recorded CDs can sound quite good with very good dacs used in very good systems. A Bruno Putzeys Mola dac might qualify as very good, but probably no Topping dac is going to be in the same league.
 
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