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Blu-ray audio not Studio Quality? |
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#1 |
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Blu-ray audio not Studio Quality?
Blu-ray audio not Studio Quality?
I have just purchased a Blu-ray player Sony BDP-S370 and as I am very interested in hi Fi classical music I also got a Blu-ray disk of a classical concert to test out the audio side of Blu-ray and the supposedly studio quality reproduction of DTS HD Master Audio. which this disc is recorded in. The video was in superb high definition and very good, the audio side did not impress me and I didn't really hear any improvement to my normal CD of the same piece of music. Investigating further, I looked to see what sort of input by Yamaha amplifier was receiving. The disc has two audio channels stereo and surround. The stereo channel was indicating PCM 48 KHZ and the surround sound 5.1 DTS-HD master 48 KHZ. I thought these figures should have been a lot higher and didn't seem to be significantly different from my other equipment______ CD PCM 44.1 KHZ. Super Audio CD, PCM 176.4 KHZ. Freeview box, PCM 48 KHZ. Satellite receiver BBC1, PCM 48 KHZ. So, from the figures above, the Super Audio CD is giving me the best reproduction and my new Blu-ray is no better than the satellite receiver and BBC 1. Is this true or what? Perhaps someone could explain the significance of these figures? |
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#2 |
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Join Date: Feb 2008
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Don't know much about the figures, only that in my experience there's no comparison between DTS HD and the BBC/satellite/freeview.
In regards to the Sony 370, you'll only get HD audio via HDMI as it doesn't support multichannel analogue outputs. So what Yamaha amp/speakers have you got, how are you connected - is it HDMI/HD audio compatible? If you are only connected via optical/coaxial you won't be getting HD audio, you'll only be getting the DTS core track, not the master track. I doubt you would hear a difference between CD/BD if connected this way. |
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#3 |
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Join Date: May 2004
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The kHz figure (only the H should be capitalised if one is being pedantic
) is the sample rate of the digital audio stream. The highest audio frequency that can be encoded is half that figure.So for a CD at 44.1kHz the highest audio frequency is 22.5kHz and at 48kHz sampling 24kHz. Teenagers and females under 30 or so might be able to hear the difference between the two but anyone male and older than 40ish is lucky to be able to hear much beyond 15-20kHz, even less if you (mis)spent your younger days with your ears pressed up against the PA stacks at rock concerts ![]() If both systems use the same number of bits per sample as well then there isn't going to be a huge difference between them. By the way the sample rate is only one factor in determining the quality of a digital stream. The bits per sample is just as important. If you sampled at 176kHz like Super Audio CD but only used 4 bits per sample it would sound a hell of a lot worse than sampling at 17.6kHz but with 24 bits per sample. Then you have to throw into the mix the various bitrate compression systems employed for broadcast and disk production. What comes out of the player or TV/Radio may well be PCM but it may not have started out as that. Digital TV uses various flavours of bitrate compression. MPEG being used for terrestrial SD broadcasts on Freeview for example. So even though the sample rate coming out of the TV/STB/PVR may be 48kHz it is in no way identical to a uncompressed 48kHz version of the same piece of audio. |
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#4 |
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Quote:
The kHz figure (only the H should be capitalised if one is being pedantic
) is the sample rate of the digital audio stream. The highest audio frequency that can be encoded is half that figure.So for a CD at 44.1kHz the highest audio frequency is 22.5kHz and at 48kHz sampling 24kHz. , the maximum frequency at 44.1KHz is only 20KHz, it's sharply filtered at that point.The idea is to be MORE than double the maximum frequency, and I would imagine that 48KHz sampling is probably limited at 20KHz as well?, but the higher the sampling frequency the less sharp the cut-off filter needs to be, and the less problems it might cause. |
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#5 |
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It can be limited (or not, if someone is daft enough to allowed aliasing) in any way the designer wishes. Sharp filters are generally ideal in frequency-domain terms, while some people prefer gentler filters for better transient response. Since the filter is acting in the ultrasonic range, it's highly debatable whether any differences are really audible.
It's quite common to use the same digital filter for a wide range of sample rates - meaning the cut off frequency (and the absolute transition bandwidth) automatically goes up if you increase the sample rate. Cheers, David. |
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#6 |
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Interesting that the OP wants to be told there are more bits on the disc - presumably so that the placebo effect causes it to sound better to him. Let's be honest: any actual extra bits over and above 48kHz 20-bit certainly won't alter the sound one jot (unless the playback equipment mangles things somehow).
Cheers, David. |
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#7 |
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That is probably true in reality but the theory behind sampling does say that the max frequency that can be sampled is half the sampling frequency. Otherwise you get into the nasty world of aliasing and the like.
Obviously it is a bit difficult to design a true brick wall filter that cuts off everything above a certain frequency absolutely perfectly. So a bit of compromise is inevitable in the real world. There is a case for saying that the effects below the cut off frequency of the filter may account for what some perceive as the "digital sound". A poorly designed filter can cause all sorts of ripples as you get close to the cut off. The steeper the fall off after that frequency the worse it can be. I doubt having a few extra kHz to play with is really going to make a huge difference But the point still stands that 44.1kHz 16bit sampling is not going to sound hugely different to 48kHz 16bit sampling all other things being equal. And having played with DAT recorders that can do both rates I can't say I have ever heard that much difference myself. |
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#8 |
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Quote:
That is probably true in reality but the theory behind sampling does say that the max frequency that can be sampled is half the sampling frequency. Otherwise you get into the nasty world of aliasing and the like.
![]() So it's not a question of it will only sample to half the sample frequency, but that it goes really badly wrong if you attempt to do so (I'm not talking just 1% extra distortion, or not even 20% it's BAD!). Quote:
But the point still stands that 44.1kHz 16bit sampling is not going to sound hugely different to 48kHz 16bit sampling all other things being equal. And having played with DAT recorders that can do both rates I can't say I have ever heard that much difference myself. |
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#9 |
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Join Date: Mar 2010
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Your speakers are probably the limiting factor, not the media format.
Do you even know if the audio was recorded at much higher resolution than that in the first place? Its a live recording, and the equipment they have will be the limit, also the setup of the mics and accoustics of the environment come into play, and finally the surround mix also matters. If its on bluray it might be a recording originally done for hdtv broadcast, who knows if they had bluray in mind. Could be they never intended for the sound quality to be anything more than broadcast quality in the first place, whether intended or for simple budgetary reasons. Its classical music after all. |
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#10 |
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Many thanks for your replies. I have a Yamaha amplifier model TSP-AX763 HDMI version 1.3A. Connected to my Sony via HDMI cable.
I was expecting to see something like this for the Blu-ray input.....96k sampling frequency / 24-bit. Not understanding PCM, I was under the impression that this was the raw digital data/bitstream coming from whatever I was playing at the time and being supplied to my amplifier for on board conversion. I see now that I was wrong and that's why everything looked the same. It looks to me that my Sony is doing all the conversion, which I don't particularly want, especially as I think the Yamaha might do a better job. If my Sony is doing all the conversion, does anybody know how to stop it and enable the raw bitstream to my Yamaha instead? Curiously, the super audio CD which shows PCM 176.4 KHZ. Which, if what you say here applies "So for a CD at 44.1kHz the highest audio frequency is 22.5kHz and at 48kHz sampling 24kHz." PCM 176.4 KHZ should give me a higher frequency response. All very strange and confusing. |
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#11 |
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Quote:
Curiously, the super audio CD which shows PCM 176.4 KHZ. Which, if what you say here applies "So for a CD at 44.1kHz the highest audio frequency is 22.5kHz and at 48kHz sampling 24kHz." PCM 176.4 KHZ should give me a higher frequency response. All very strange and confusing.
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#12 |
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Join Date: Mar 2010
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Sacd works in a completely different way than cd, you cannot do direct comparison from a single figure. google up the wiki
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#13 |
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Join Date: Feb 2008
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Quote:
Many thanks for your replies. I have a Yamaha amplifier model TSP-AX763 HDMI version 1.3A. Connected to my Sony via HDMI cable.
I was expecting to see something like this for the Blu-ray input.....96k sampling frequency / 24-bit. Not understanding PCM, I was under the impression that this was the raw digital data/bitstream coming from whatever I was playing at the time and being supplied to my amplifier for on board conversion. I see now that I was wrong and that's why everything looked the same. It looks to me that my Sony is doing all the conversion, which I don't particularly want, especially as I think the Yamaha might do a better job. If my Sony is doing all the conversion, does anybody know how to stop it and enable the raw bitstream to my Yamaha instead? Curiously, the super audio CD which shows PCM 176.4 KHZ. Which, if what you say here applies "So for a CD at 44.1kHz the highest audio frequency is 22.5kHz and at 48kHz sampling 24kHz." PCM 176.4 KHZ should give me a higher frequency response. All very strange and confusing. |
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#14 |
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Join Date: Jan 2003
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It is not just the sampling rate but the bit rate that makes a big difference between bog standard DTS and DTS Master audio. On my rig there is no comparison when watching movies, DTS is good but DTS MA is unbelievable.
It could also just be that it is not a very good sound mix. And regardless of whether your player is sending the raw PC or your amp is out putting the DTS MA there should be no difference in quality. The PCM you out put of HDMI is the raw signal your Blu ray player will not be doing nay conversion (this only happens if you use analogue outs to a non HD amp). |
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#15 |
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Quote:
Have a look at page 20 in the manual - from what I can make out you need to set the BD audio MIX setting to off, this should pass the primary soundtrack to the amp for decoding.
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#16 |
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Quote:
Many thanks, but I have already tried that and it was set to off, I have also tried it set to on but for either setting it still shows PCM 48 KHZ.
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#17 |
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most reviews of amps/receivers also indicate that audio reproduction through a 5.1 system, is rarley as competent as a dedicated and probably equally expensive stereo amp.
it stands to reason, really as there is so much more stuff in a 5.1 receiver, they must have cut corners somewhere |
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#18 |
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Ob my amp a DTS-HD Master Audio soundtrack shoes as 96 kHz/24bit resolution. My amp Denon AVR-3310 also shows DTS-HD as the input source as well.
One question, though, how have you connected you Blu-ray player to the amp? I have a Sony BDP-S350 and HD audio is only available on the HDMI output, as the BDP-S350 doesn't have an internal HD audio decoder so the co-ax and optical audio outputs don't have HD audio output capability. I also had to configure the sound output to HDMI. Then the amp does the sound decoding for the HD audio. |
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#19 |
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[QUOTE
One question, though, how have you connected you Blu-ray player to the amp? [/quote] Many thanks for your replies. I have a Yamaha amplifier model TSP-AX763 HDMI version 1.3A. Connected to my Sony via HDMI cable. |
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#20 |
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Quote:
One question, have you switched DSD output mode on, as this enables bitstream for SACD...?
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#21 |
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The threads gone on this long without us finding out what this bluray disc was?
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#22 |
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Join Date: Feb 2008
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Quote:
Many thanks, but I have already tried that and it was set to off, I have also tried it set to on but for either setting it still shows PCM 48 KHZ.
Are you getting DTS HD showing up on the amps main display? If you are, then the amp will be doing the decoding and you will be getting the best soundtrack quality available. Whether the Yamaha will be better at decoding than the player I don't know. I use a PS3 that will only pass LPCM over HDMI and I can't tell a difference between this and other amps doing the decoding via bitstream. |
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#23 |
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Quote:
To explain this a little more clearly - you must NOT allow any frequencies higher than half the sampling rate to be digitised - it's really, really bad news (or at least sound) if you do
![]() So it's not a question of it will only sample to half the sample frequency, but that it goes really badly wrong if you attempt to do so (I'm not talking just 1% extra distortion, or not even 20% it's BAD!). I don't as there is any difference, and as I said both probably use the exact same 20KHz filtering. Is it fair to say that frequencies above half the sampling rate reappear as reflections going back down the spectrum? ie with a 48k system, a component at 25kHz would appear as an unwanted 23kHz distortion. Or is it much worse than that and in this scenario a 1kHz output would result?
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#24 |
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Quote:
This is something I still can't quite get my head around.
Is it fair to say that frequencies above half the sampling rate reappear as reflections going back down the spectrum? ie with a 48k system, a component at 25kHz would appear as an unwanted 23kHz distortion. Or is it much worse than that and in this scenario a 1kHz output would result? ![]() Basically you MUST block all frequencies below half the sampling rate, blocking even lower is better still - which is why they use higher samplig rates. |
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#25 |
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Much, much worse than that - you get random (or almost random) low frequencies, depending on the exact relationship between the sampling frequency and the audio.
Basically you MUST block all frequencies below half the sampling rate, blocking even lower is better still - which is why they use higher samplig rates. Just found some info on Wiki which hopefully helps. http://en.wikipedia.org/wiki/Aliasing#Folding |
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All times are GMT. The time now is 17:07.



) is the sample rate of the digital audio stream. The highest audio frequency that can be encoded is half that figure.