Dolby's TrueHD 96K Upsampling To Improve Sound On Blu-Rays

Stowie101 writes in with a story about your Blu-ray audio getting better. "The audio on most Blu-ray discs is sampled at 48kHz. Even the original movie tracks are usually only recorded at 48kHz, so once a movie migrates to disc, there isn't much that can be done. Dolby's new system upsamples that audio signal to 96kHz at the master stage prior to the Dolby TrueHD encoding, so you get lossless audio with fewer digital artifacts. The 'fewer digital artifacts' part comes from a feature of Dolby's upsampling process called de-apodizing, which corrects a prevalent digital artifact known as pre-ringing. Pre-ringing is often introduced in the capture and creation process and adds a digital harshness to the audio. The apodizing filter masks the effect of pre-ringing by placing it behind the source tone — the listener can't hear the pre-ringing because it's behind the more prevalent original signal."

Comments

You cant hear it anyway.

[Hatta]

44.1khz audio is already transparent to the human ear. Blind studies have been done where a 16 bit 44.1khz ADC-DAC pair was inserted into a high resolution analog audio source. No significant difference was observed.

Don't waste money on the placebo effect.

Re:You cant hear it anyway.

[Anonymous Coward]

The hearing limit is actually about 20kHz. You need more than 40kHz sampling if you want to capture a sine wave at 20kHz.

The purpose of capturing at higher than 48kHz is to prevent sounds at frequencies above 20kHz being captured at a too-low sampling frequency, and appearing as audible frequencies. These can be removed by analog filtering, but only about one octave above the cutoff frequency. Analog filters are not ideal brick-wall filters, so 96kHz sampling is useful.

However, once the audio is acquired and digitized, software can provide a true brick-wall digital filter. This is impossible to do in analog hardware. After applying the brick wall filter, it can be sampled down to 48kHz or 44kHz with no loss. So, there is absolutely no reason to put 96kHz on disc.

The article isn't clear whether it's 96kHz on just the master, or the disc also.

Re:You cant hear it anyway.

[SimonTheSoundMan]

I'm a sound engineer and you are totally right.

Going back in history. 44.1kHz was chosen because it syncs with PAL video frames, 48kHz syncs with NTSC. If you were doing linear editing, you can dub and cut the audio perfectly to the half frame.

44.1kHz stuck because Umatic, an analogue videotape that you could buy a PCM head as an optional extra, was chosen to create the master copies for CDs to be sent to duplication in to pressed CDs.

Re:You cant hear it anyway.

[dgatwood]

Actually, polystyrene caps can make a huge difference over electrolytic or tantalum caps in certain parts of some circuits. For example, in condenser microphones, the coupling capacitor between a microphone element and the first FET stage is a critical part of the circuit in which the signal level is very, very weak. Thus even tiny amounts of noise from cheap capacitors can have a significant effect on the final result. A fair number of cheap Chinese microphones sound dramatically better if you replace the cheap dipped tantalum caps they use with a film cap or poly cap.

We're not talking about a small difference here, either. We're talking night and day. A deaf person could just about hear the difference. :-) Replacing just the handful of tantalum capacitors in those microphones can make the difference between a muddy sound with a harsh, brittle top end and a fairly clean, accurate representation of what is being recorded... all for about five bucks and a few minutes of soldering. (Even better, the most important one—the FET coupling cap—is usually direct-wired between the capsule mount and the FET's lead, so you don't have to worry about lifting traces....)

Capacitors within the feedback path of an amplifier circuit can also degrade the sound noticeably. Admittedly, this isn't as much of an issue these days with the rise of modern, chip-based amplifier circuits, but it is still worth keeping in mind, particularly given that most condenser microphones still use transistor-based amplifier circuits.

Just to be clear, though, it doesn't have to be polystyrene film. The difference between a polystyrene cap and a traditional metal (polyester) film cap is negligible compared with the difference between film caps and electrolytic or (*shiver*) tantalum caps. Tantalum caps simply should not be within a city block of any trace that carries an audio signal.... Okay, slight exaggeration, but you get my point.

And, of course, it doesn't make sense to replace every capacitor. If it isn't in the signal path, it usually won't make much difference (though the absence of capacitors in the right places on power supply rails can cause some fun problems), and even if it is, it may or may not make much of a difference, depending on where the capacitor is in the signal path.

Apodizing Filter

[Josuah]

The title is misleading if the actual goal of this is to apply an apodizing filter. I suspect the reason it's called "Advanced 96K Upsampling" is because that's much easier to get people to buy into that than a "Apodizing Filter" sticker.

The article explains how the audible benefit comes from the application of the Meridian apodizing filter, which changes the analog signal reproduced from digital data by reducing the pre-ringing. IIRC the trade-off is that post-ringing increases. The claimed benefit is that since the ringing now occurs after the "real" music of larger amplitude and as a result the ringing is masked or could be considered like an acoustic echo that naturally occurs.

The 96K upsampling is just a side-effect of wanting the extra samples when you are applying the filter.

Here's a decent summary of what is supposed to happen to the analog audio signal as a result of the filter application: Technical analysis of the Meridian Apodizing filter.

That being said, from what I've read over the past few years I think people are kind of mixed on whether or not the filter makes things better, worse, or just different but not better.

Re:You can prefectly represent anything up to Fs/2

[Man On Pink Corner]

Try a high but more audible frequency.

It may be less confusing if I put it this way: If you can't hear a sine wave beyond, say, 20 kHz, then you are not going to be able to tell the difference between a sine wave at 7 kHz and a square wave whose fundamental frequency is 7 kHz. That's because the lowest harmonic in the square-wave signal will be at 21 kHz. Your ears will filter it out, just as the antialiasing filter in the recording system would need to do.

Now, that being said, the argument has been made that intermodulation effects in the human ear can allow us to perceive sounds beyond the usual 20 kHz limit when they mix with each other. To the extent these effects occur when listening to the source material at a given level, you could argue that the ultrasonic parts of a performance should be captured and reproduced along with everything else, and that would require a higher sampling rate.

The showstopper for this argument is that any desirable sonic content resulting from IMD at ultrasonic frequencies could only be reproduced "properly" at a specific volume level, because distortion products by definition are generated by nonlinear processes.

And Harry Nyquist is rolling around in his grave

[Gordo_1]

Not that this whole thing isn't absurd for the reasons already discussed above, but what no one bloody well seems to understand it that an audio stream is not a godamn bitmap picture. You can't improve audio quality of *audible frequencies* by increasing resolution of the horizontal axis (sampling frequency) beyond a rate which surpasses the Nyquist frequency for human hearing. Assuming a high quality anti-aliasing filter is used and excellent quality recording and playback equipment, audio sampled at 48kHz can be unambiguously represented up to about 24kHz. 96kHz is a waste of bits.

Vertical resolution (# of bits) is the only theoretical way to improve actual audio quality further... and beyond about 16-18 bits, it's also beyond the ability of even the most diehard audiophiles to discern (in properly conducted experiments.)

Re:Worthless gimmick with no audible benefits

[Anonymous Coward]

The signal is only "stair stepped" because they chose to graph it that way. The audio signal coming out of the DAC does not look like that. Those stair-steps are happening at frequencies more than half the sampling rate -- they are eliminated from the analog output by a low-pass filter. This is essentially performing this "splining" you are talking about.