24/192 Streaming Audio Playback via Marantz AVR

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classic_erik

Enthusiast
Will the Marantz SR7015 DAC compress 24/192 Amazon Ultra HD streaming music via the HEOS app down to 48 kHz even if I use Pure Direct mode bypassing Audyssey? Same question applies if I use a BlueSound NODE or Cambridge CXN V2 to stream Tidal MQA instead of Amazon Ultra HD via HEOS. Basically, does the AVR’s DAC create a impenetrable wall to 24/192 playback even if Audyssey is disabled? Since there is no way to validate the actual bitrate in the Marantz AVR I guess we will never know?
 
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Golfx

Full Audioholic
As i recall Amazon Music via HEOS displays its bit rate from Denon/Marantz AVRs on your TV. The Heos app is just to control upcoming choices. The music itself streams via internet/Wi-Fi from amazon’s servers—not your phone app. And if you shut off audyssey it will not downgrade your clock speed to 48Khz but leave it at 24/192. Which you can check using the info button on your remote.
 
C

classic_erik

Enthusiast
As i recall Amazon Music via HEOS displays its bit rate from Denon/Marantz AVRs on your TV. The Heos app is just to control upcoming choices. The music itself streams via internet/Wi-Fi from amazon’s servers—not your phone app. And if you shut off audyssey it will not downgrade your clock speed to 48Khz but leave it at 24/192. Which you can check using the info button on your remote.
Isn’t the bit rate displayed on the HEOS screen just the signal rate rather than what is actually being output by the AVR? Also when I click the info button I don’t see a bit rate, only the signal and output formats.
 

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PENG

Audioholic Slumlord
Will the Marantz SR7015 DAC compress 24/192 Amazon Ultra HD streaming music via the HEOS app down to 48 kHz even if I use Pure Direct mode bypassing Audyssey? Same question applies if I use a BlueSound NODE or Cambridge CXN V2 to stream Tidal MQA instead of Amazon Ultra HD via HEOS. Basically, does the AVR’s DAC create a impenetrable wall to 24/192 playback even if Audyssey is disabled? Since there is no way to validate the actual bitrate in the Marantz AVR I guess we will never know?
That's some old questions and the old answers should apply, such as can be found in the following:

Audyssey MultEQ Room Correction Interview With Chris Kyriakakis | Audioholics

below are the relevant parts:

Audioholics:
What is the default frequency range corrections are applied to? In other words, is there a frequency ceiling or floor above/below which correction isn't applied? If correction isn't applied full band, please explain.

Chris Kyriakakis: MultEQ is capable of applying correction from 10 Hz to 24 kHz. However, during the measurement process it first determines the roll off points of each speaker and subwoofer, and limits the correction below that point.


Audioholics: The top frequency for correction is 24kHz, implying that Audyssey is functioning at a 48kHz sample rate. Does this mean that high resolution content (for example 192kHz or 96kHz sample rate PCM) will be downmixed?

MultEQ is capable of applying correction from 10 Hz to 24 kHz.
Chris Kyriakakis: There are two parts to this answer. A loudspeaker does not reproduce acoustic energy above about 24-30 kHz even if it was in the content (with the exception of super-super tweeters), and a microphone cannot capture acoustic energy above that range. So if there is no information captured then, there is nothing for the filter to do up there.

Now, there is content encoded at higher sampling rates of course. We offer MultEQ at 96 kHz and even higher if needed so that the content can be processed without downsampling, even though the MultEQ filters above 24-30 kHz (adjustable) would be doing absolutely nothing. The issue is that doubling the sampling rate also doubles (roughly) the processing requirements needed. This is true for any kind of digital processing not just MultEQ. The AVR makers would have to add significant cost for more DSP processing and they have chosen not to do that. So they decided to use Audyssey at a max of 48 kHz. From an acoustic point of view this makes perfect sense for the reason I explained above.
 
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PENG

Audioholic Slumlord
Will the Marantz SR7015 DAC compress 24/192 Amazon Ultra HD streaming music via the HEOS app down to 48 kHz even if I use Pure Direct mode bypassing Audyssey?
Downsamping to 48 kHz from 96 or 192 kHz is not "compress", just down sample, not the same..

Also, as an aside, down sampling, as long the sampling frequency is at or above 44.1 kHz, there will be no loss in information within the audio band (CD specs) that only goes up to 20 kHz. 44.1 kHz will ensure frequencies up to 22.05 kHz are preserved per Nyquist. Regardless of what you might have read on the internet, the following is factual.

Nyquist frequency - Wikipedia
Nyquist frequency
From Wikipedia, the free encyclopedia

In signal processing, the Nyquist frequency (or folding frequency), named after Harry Nyquist, is a characteristic of a sampler, which converts a continuous function or signal into a discrete sequence. In units of cycles per second (Hz), its value is one-half of the sampling rate (samples per second).[1][2][A] When the highest frequency (bandwidth) of a signal is less than the Nyquist frequency of the sampler, the resulting discrete-time sequence is said to be free of the distortion known as aliasing, and the corresponding sample rate is said to be above the Nyquist rate for that particular signal.[3][4]
In a typical application of sampling, one first chooses the highest frequency to be preserved and recreated, based on the expected content (voice, music, etc.) and desired fidelity. Then one inserts an anti-aliasing filter ahead of the sampler. Its job is to attenuate the frequencies above that limit. Finally, based on the characteristics of the filter, one chooses a sample rate (and corresponding Nyquist frequency) that will provide an acceptably small amount of aliasing.
In applications where the sample rate is pre-determined, the filter is chosen based on the Nyquist frequency, rather than vice versa. For example, audio CDs have a sampling rate of 44100 samples/sec. The Nyquist frequency is therefore 22050 Hz. The anti-aliasing filter must adequately suppress any higher frequencies but negligibly affect the frequencies within the human hearing range; a filter that preserves 0–20 kHz is more than adequate for this.

Your question on pure direct:

If you use direct or pure direct mode, Audyssey would be disabled anyway, so there should be no down sampling. If I remember right, the manuals don't see anything about this but it is known from online reviews/measurements.
 
C

classic_erik

Enthusiast
Ok sorry yes I was referring to down sampling; thank you for the correction.
My initial question was whether the AVR’s DAC would <down sample> the signal down to 48 kHz even with Audyssey turned off, but you are search shaming me with an irrelevant post about Audyssey rather than offering a helpful suggestion.
 
P

PENG

Audioholic Slumlord
My initial question was whether the AVR’s DAC would <down sample> the signal down to 48 kHz even with Audyssey turned off, but you are search shaming me with an irrelevant post about Audyssey rather than offering a helpful suggestion.
Sorry I missed that part of your question. The AVR's DAC would not down sample, the IC used was the AK4458 originally and since the spring of 2021 (according to D+M) they replaced it with another IC and it was rumored to be the PCM5102A that is not as good, but it is still capable of up to 384 kHz.

PCM510xA 2.1 VRMS, 112/106/100 dB Audio Stereo DAC with PLL and 32-bit, 384 kHz PCM Interface datasheet (Rev. C) (ti.com)

So if Audyssey is tuned off, digital stereo (2 channel) signals with sampling rate up to 192 kHz won't get down sampled, there is no need to.
 
C

classic_erik

Enthusiast
Sorry I missed that part of your question. The AVR's DAC would not down sample, the IC used was the AK4458 originally and since the spring of 2021 (according to D+M) they replaced it with another IC and it was rumored to be the PCM5102A that is not as good, but it is still capable of up to 384 kHz.

PCM510xA 2.1 VRMS, 112/106/100 dB Audio Stereo DAC with PLL and 32-bit, 384 kHz PCM Interface datasheet (Rev. C) (ti.com)

So if Audyssey is tuned off, digital stereo (2 channel) signals with sampling rate up to 192 kHz won't get down sampled, there is no need to.
Cool thanks that’s what I was looking for. Much appreciated.
 
lovinthehd

lovinthehd

Audioholic Jedi
I see this bit in Heos info:

HEOS Built-in lets you stream from the world’s leading music platforms, including Spotify, Deezer, TIDAL, Napster, TuneIn, Mood Mix, and Amazon Music HD. Plus, it lets you stream in high resolution, and while that currently means anything between 16-bit/ 44.1kHz and 24-bit/192kHz, HEOS Built-in is already equipped to go higher.

This in the avr manual
Supported audio formats
2-channel Linear PCM - 2-channel, 32 kHz – 192 kHz, 16/20/24 bit
Multi-channel Linear PCM - 7.1-channel, 32 kHz – 192 kHz, 16/20/24 bit
 
Trell

Trell

Audioholic Spartan
I see this bit in Heos info:

HEOS Built-in lets you stream from the world’s leading music platforms, including Spotify, Deezer, TIDAL, Napster, TuneIn, Mood Mix, and Amazon Music HD. Plus, it lets you stream in high resolution, and while that currently means anything between 16-bit/ 44.1kHz and 24-bit/192kHz, HEOS Built-in is already equipped to go higher.

This in the avr manual
Supported audio formats
2-channel Linear PCM - 2-channel, 32 kHz – 192 kHz, 16/20/24 bit
Multi-channel Linear PCM - 7.1-channel, 32 kHz – 192 kHz, 16/20/24 bit
You actually went to the manual looking for information? :eek:
 
C

classic_erik

Enthusiast
Thanks @lovinthehd

Anyone know of a way to actually confirm the playback bitrate and sampling rate on a Marantz AVR? If the signal says 24/192 in HEOS do I just assume that’s what is being output?
 
lovinthehd

lovinthehd

Audioholic Jedi
If the indication is its a 24/192 signal and you're not using Audyssey....why worry about it?
 
C

classic_erik

Enthusiast
If the indication is its a 24/192 signal and you're not using Audyssey....why worry about it?
Not particularly worried; just like to validate my setup for any content so was wondering if there was a way.
 
Trell

Trell

Audioholic Spartan
If the indication is its a 24/192 signal and you're not using Audyssey....why worry about it?
My guess is that she is unsure whether or not this is just an indication of incoming signal that is then subsequently down converted. To be honest, not unreasonable at all, but won't matter audibly.
 
lovinthehd

lovinthehd

Audioholic Jedi
Not particularly worried; just like to validate my setup for any content so was wondering if there was a way.
I'd say the indication of 24/192 is the best you'll get. I understand wanting to get what you're "paying" for, but it's overrated to begin with.
 
P

PENG

Audioholic Slumlord
Thanks @lovinthehd

Anyone know of a way to actually confirm the playback bitrate and sampling rate on a Marantz AVR? If the signal says 24/192 in HEOS do I just assume that’s what is being output?
Aside from measurements such as those by ASR, I don't think there is a way, I don't think there is a way because the output signal of the DAC is an analog signal, that is, no more bit depth/bit rate (aka sampling rate) to display anyway.

If you look at the signal flow diagrams in the Marantz service manual:

1662237477770.png

You can see that the digital signal will be converted to analog by the DAC and then go straight to the volume control IC, then the HDAM is the final output. The above is for the AV8805, the newer AVRs from the SR6014 through SR7015 and likely theSR8015 too, may have another Opamp buffer after the HDAM for the final output (pre out). From the DAC output on, the signal is in analog form, there is no bit depth and/or sampling rate to display.

That's why as you said, it will only display the sampling rate of the input signal.

Obviously the other indirect way is to trust Marantz who says:

Aside from bench test measurements such as those done by ASR, you will have to trust what Marantz says in the owner's manual.

Pure Direct This mode plays back an even higher quality sound than the “Direct” mode.
The following circuits are stopped in order to further improve sound quality.
* Display indicator circuit of the main body (display will go off.)
* The analog video input/output switcher and processor is disabled.
As the DSPs are not used, there won't be down sampling, as HD said, as long as you use pure direct mode of even direct mode, there will be no downsampling and the DAC will just convert the digital input signal "as is" at whatever bit depth/sampling frequency such as 24bit/192 kHz, if the input signal is at higher sampling rate than 192 kHz, than it just will not be able to play the file, but I am sure you know that already.
 
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