Player decoding or Receiver decoding, that is the question

[EWL5]

Originally Posted by welwynnick

I don't know what's different about this forum, but many people seem to take their opinons from those who claim to know what they're talking about, rather than being a bit more scientific (which I am!) and learning and understanding things for themselves. Unfortunately this is a complicated question that I've been endeavouring to unravel for a year or two, and I've only made some practical progress in the last few days. I've been reporting things on a day-by-day basis elsewhere, so there's a documented record, and this is quite pertinent.

Firstly, there's no reason why bitstream should sound the same as LPCM. The digital audio replay architecture is different, and that means the sound may be different. I'd better explain that, quick.

A digital audio signal has two elements - a data stream and a timing stream. Both of these end up at the DAC to create the analogue audio signal, and both need robustness and fidelity.

These streams are quite distinct, and take different critical paths:
The data stream starts at the disc, and ends up at the DAC.
The timing stream starts at the master clock, and also ends up at the DAC.

The bitstream decoding process only affects the data stream, and I think we are all pretty confident that that is lossless, so the DAC is receiving the right data, whichever decoding architecture is used.

However, this architecture MAY affect the path of the timing stream . This isn't necessarily the case, but my experience with the Onkyo suggests that it is. With LPCM, the clock is in the transport, and the path takes the HDMI connection to the receiver and too the DACs. This is a tortuous path, and is full of degradation - like toslink but worse. With bitstream, there's no clock associated with the data because its compressed, and the clock is regenerated in the amplifier, just as it is with DD and DTS. This places the clock right next to the DAC, with robust and direct connection, which is ideal. This doesn't eliminate interference, noise and jitter, but its a very big help.

So with bitstream there is an opportunity to minimise jitter, though this is only achieved if the amplifier doesn't generate its audio clock from the players video clock, in the same way it does with LPCM. For ayear, I've never known which it was, but I think that question has now been answered in my mind, now. Of course, my observations may not apply to other amplifiers, like Pioneers, which have different architectures. I'm itching to find out....

Nick

I'd love to do a double blind jitter test with you. I haven't met a person who claims to hear jitter during multichannel movie soundtracks. With the dynamics involved, it's almost pointless (perhaps a different story for high resolution stereo music).

[welwynnick]

Originally Posted by EWL5

Guys, you may be interested in the following thread I started in AVS:
http://www.avsforum.com/avs-vb/showt...1066645&page=1
It is double blind and I had FilmMixer (soundtrack mixer insider) agree with me concerning the results.
Edit: Anybody feel it would be a good idea for me to copy the first post over to Blu-ray.com either as a new thread or to an existing discussion?

Very interesting thread to read. Having experienced it myself, I'm full of admiration for anyone who can put these sorts of tests together and open themselves up to scrutiny by people who have made no effort to contribute to the greater good. It really is hard work, especially when you keep pulling favours, breaking your back over invisible connections, and you make yourself look like an idiot.

Our tests are slightly crossed. You were concentrating on amplitude data degradation, and I was concentrating on timing degradation, so not quite the same thing. As a matter of interest, I used to have an LPCM-capable system, but I've been relying on DTS re-encode from an LG BH200 for my BD playback. It's a great idea and its does sound good, but I always thought it wasn't the real thing.

I'm well aware of Filmmixers longheld views, and though I think he's a genuine and reliable contributor, I don't believe him. We came up with the following order of sound quality for some of the configurations that I tried:

1. 3800 or BH200 to 886 by bitstream
2. 3800 to MC12 by analogue
3. 3800 to 886 by analogue
4. 3800 to 886 by LPCM
5. BH200 to 886 by LPCM
6. BH200 to MC12 by DTS re-code
7. BH200 to 886 by DTS re-code
8. BH200 to 886 by analogue (presumably!)

A couple of them were perhaps difficult to be sure about, like 5. & 6. Please note that some people claim to hear the difference between bitstream from different players, or even between spdif & toslink, but I don't! I don't think I have golden ears at all.

Originally Posted by EWL5

I'd love to do a double blind jitter test with you. I haven't met a person who claims to hear jitter during multichannel movie soundtracks. With the dynamics involved, it's almost pointless (perhaps a different story for high resolution stereo music).

I know a lot of respected people who think the same thing, but I don't agree. Perhaps jitter has more impact on music, but I find it noticable with films as well. The first time I heard bitstream I shall never forget: it was chapter 1 of the Golden Compass, on the Denon 3800, so a good player and film, not music, but the difference was still quite apparent.

This is what I listen to: Bitstream, like other jitter-reducing methods, gives much the same sound, but better resolution and imaging. Its especially good at retaining the tiny, vanishing, decaying notes and echos that analogue audio was always good at. I think this has a number of advantages:

• You get better focus on each subject.
• There is greater depth in the image.
• You can hear musical instruments playing continuously and independently of everything else - you hear them concurrently, not sequentially.
• Musical instruments become joined-up and coherent, instead of independant and unrelated.
• The timing of music is improved, because the gaps between notes are filled with delicate decays and reverberations.
• They were there on purpose when the piece was produced, played and recorded.
• Without this small-signal fidelity, you just hear sound - pause - sound - pause etc, which is empty and boring, there's no flow, pace or variety.

These were the things that I was looking for (I hope some of you at least understand what I'm prattling on about) and I found it quite easy to hear the differences. These differences were also quite apparent with all soundtrack, not just with music!

BR, Nick

[EWL5]

Originally Posted by welwynnick

Very interesting thread to read. Having experienced it myself, I'm full of admiration for anyone who can put these sorts of tests together and open themselves up to scrutiny by people who have made no effort to contribute to the greater good. It really is hard work, especially when you keep pulling favours, breaking your back over invisible connections, and you make yourself look like an idiot.

Our tests are slightly crossed. You were concentrating on amplitude data degradation, and I was concentrating on timing degradation, so not quite the same thing. As a matter of interest, I used to have an LPCM-capable system, but I've been relying on DTS re-encode from an LG BH200 for my BD playback. It's a great idea and its does sound good, but I always thought it wasn't the real thing.

I'm well aware of Filmmixers longheld views, and though I think he's a genuine and reliable contributor, I don't believe him. We came up with the following order of sound quality for some of the configurations that I tried:

1. 3800 or BH200 to 886 by bitstream
2. 3800 to MC12 by analogue
3. 3800 to 886 by analogue
4. 3800 to 886 by LPCM
5. BH200 to 886 by LPCM
6. BH200 to MC12 by DTS re-code
7. BH200 to 886 by DTS re-code
8. BH200 to 886 by analogue (presumably!)

A couple of them were perhaps difficult to be sure about, like 5. & 6. Please note that some people claim to hear the difference between bitstream from different players, or even between spdif & toslink, but I don't! I don't think I have golden ears at all.

I know a lot of respected people who think the same thing, but I don't agree. Perhaps jitter has more impact on music, but I find it noticable with films as well. The first time I heard bitstream I shall never forget: it was chapter 1 of the Golden Compass, on the Denon 3800, so a good player and film, not music, but the difference was still quite apparent.

This is what I listen to: Bitstream, like other jitter-reducing methods, gives much the same sound, but better resolution and imaging. Its especially good at retaining the tiny, vanishing, decaying notes and echos that analogue audio was always good at. I think this has a number of advantages:

• You get better focus on each subject.
• There is greater depth in the image.
• You can hear musical instruments playing continuously and independently of everything else - you hear them concurrently, not sequentially.
• Musical instruments become joined-up and coherent, instead of independant and unrelated.
• The timing of music is improved, because the gaps between notes are filled with delicate decays and reverberations.
• They were there on purpose when the piece was produced, played and recorded.
• Without this small-signal fidelity, you just hear sound - pause - sound - pause etc, which is empty and boring, there's no flow, pace or variety.

These were the things that I was looking for (I hope some of you at least understand what I'm prattling on about) and I found it quite easy to hear the differences. These differences were also quite apparent with all soundtrack, not just with music!

BR, Nick

Nick,

This is my current stance:

1) Lossless audio over lossy audio (found in BD, not the lower bitrate lossy found on DVD, which is more of a difference) is not that different. On this point we disagree.

2) Decoding in the AVR/prepro is superior to decoding in the player, but NOT primarily because of jitter. Go back to this post to see the reasons why I feel the AVR/prepro has the better tools for the job:

https://forum.blu-ray.com/showpost.p...4&postcount=57

[welwynnick]

Originally Posted by EWL5

Nick,

This is my current stance:

1) Lossless audio over lossy audio (found in BD, not the lower bitrate lossy found on DVD, which is more of a difference) is not that different. On this point we disagree.

2) Decoding in the AVR/prepro is superior to decoding in the player, but NOT primarily because of jitter. Go back to this post to see the reasons why I feel the AVR/prepro has the better tools for the job:

https://forum.blu-ray.com/showpost.p...4&postcount=57

Thats a great post, much more meaningful than most. I have to duck out now, but briefly, I tried to eliminate such varibles by having bass management, time alignment and equalisation all switched off. I just had the speaker levels matched

One final comment, the Denon and the Onkyo actually have the same DACs, but the analogue sound quality from the Denon, in as far as it could be isolated from the Onkyo, was rather better.

One final test I did was to briefly connect the Denon analoge outputs directly to the amplifier, and I found another veil was lifted. It was sounding really excellent.

Nick