[Reclock Pot Player 64bit 11
Virginie Fayad
There is a lot of interests on the Innuos and its capability of reclocking the USB before output to a connected DAC, I am not sure if its SQ is comparable to HQPlayer? Has anyone here any experience with the Innuos products?
Yes, everything works fine and sounds wonderful with the ethernet in, but i would like to explore another venue with the hqplayer because it has a good name, also i am curious to see if it does make any differences or even sounds better.
Note that the latest Linn Klimax DSM/3 supports copper ethernet, optical ethernet, USB B, toslink and wifi. Obviously, I do not use wifi because it is not efficient and potentially unstable, even with the latest wifi mesh technology.
Thanks @jussi_laako. However, my question is wheter you would recommend any USB reclocking device (for example the Innuos Phoenix USB) after the HQPlayer output? Or using any HQPlayer NAA enabled device?
Going back to your original question: like @Jez and @jussi_laako I see no benefit in throwing in the Innuos USB reclocker in front of the DSM/3. Your Mac Mini setup should allow you to explore what HQPlayer can do for your system.
1. Download reclock here: -Filter
2. Install reclock.
3. Open the "Configure Reclock" shortcut that is on the desktop after reclock installation.
4. Check the "when speeding up" checkbox, and press ok.
5. In Zoom Player , under advanced options / playback / audio , set "Reclock audio renderer" as the audio renderer.
Then, you only need to press shift+z to have Zoom Player play fast with normal sound pitch.
You should be able to do this I think. You can launch Zoom with command line parameters, one of which is ExFunc, which allows you to use one of the extended functions. One of those functions is exSetPlayRate, which allows you to set the play rate. The other possibility is to edit the skin you are using to make a button to call that exact same extended function (I'm not sure if there is a skin already in the skins gallery that has a built-in button for play rate).
So, I finally figured it why I got frame drops, it's cause the video don't get synced right with the audio. Every time I play a video I have to do it manually, go to the tray icon for reclock and change to the frame rate, the video is for. Doing this gets me no dropped frames, at all, throughout the whole video, and this is while even using MadVR. Could someone please, help me find a permanent solution, so this can be done automatically?
When I play video files, sometimes this ReClock launches (Even when SVP is not launched)
and asks me if I want to use it with my Windows media player. As far as I know, this only
shows when I try to play video file with my WMP.
I remember I've DL'd some kind of program that was introduced in this forum, and I think
that's what it's causing this. But I can't remember the name. I can't find any thing suspicious
on list of "Uninstall program" at Control panel.
I've got the opposite problem in that ReClock now never nags me on opening files (it used to, on first install of SVP 3.1.7) and doesn't appear to work at all, which sucks as I believe it could fix my issue with audio lag in D3D Fullscreen mode.
Mashingan, yeah it's already set to "Never Load" DX
I think I gotta find out the name program I've DL and installed from this forum
some months ago to fix my problem, and I cannot remember what's the name it was
and what exactly does DX
If I use "Set ReClock as preferred renderer (not recommended)" then ReClock detects that the video is being played at a fluctuating 59.9XX FPS via SVP (even though it says "No Video Stream Found") but reports that it is doing nothing to the video, is altering the audio and its icon is red. ??
I have a modded Esoteric D70 DAC, which I like very much. However, to take advantage of its dual word clock capability I would need to purchase something like the Weiss AFI1 interface. An alternative would be to use a reclocker, like the Empirical PaceCar. Another alternative would be to swap this DAC for another that combines the two, such as a Weiss 202 or Antelope Zodiac+.
Steve Kuh[br]Mac Mini > Glyph HD > Weiss AFI1 (slave) > modded Esoteric D70 (master) > BAT VK51SE > Classe CA400 > Harbeth Super HL5[br]\"Come on the amazing journey and learn all you should know...\"
I was running some tests with a DVD-Audio player via spdif output into my L22 audio card and analog output from the card to my processor. This was sounding pretty great and I was wondering why as the computer usually sounds better:) I entered some commands and found one that would make it sound bad again:) I called Lynx and they advised that my initial pass through was being reclocked by the L22 card and the subsequent command would turn off the reclocking feature. WOW what a difference the reclock makes!
"In the D70, a servo-free direct clocking system is used, which comes close to the ideal of control. In systems where a clock other than the one derived from the digital audio signal is used, typically PLL technology is used to keep the clock frequency as constant as possible.
* * * The new method employed in the D70 to maximize audio quality is to use a DAC locked to a direct crystal-controlled fixed-frequency clock. This allows the conversion to be carried out independently from the input frequency and input jitter is therefore eliminated. In the case of non-synchronization, the RAM Link buffer eliminates any potential problems."
You have a audio source in 44.1Khz/16bit. Its digitally modulated in PCM. Now it gets read. Its pulse modulated to your transfer-media. Let it be Ethernet, USB, SPDIF, whatever, it doesn't matter. To get that stream modulated over the cable/fibre it needs to went through the multiplexing units on the interfaces. Then multiplexed on the cable. Demultiplexed again. Demodulated. And then fed directly without any further digital processing to the DAC. Means: 44.100 ticks per second 16 bit big data segments. Segment for segment it needs to be as accurate as 0.00002267573696 second. No segment has to arrive faster or slower than after 0.00002267573696 second. If it would the sound couldnt be reconstruected as it was.
Now there are resampling DACs which use a not-100% accurate rate and resample it to a accurate one. What has to be done for this? Sure it has to buffer. Wait. Wait. Wait. Calculating the rate and finding it out of the average rate of the receiving data stream. Now it has enough and can stream this data clocked as precise as possible with a local clock to the final converting DAC chip.
There is no other good way to do this and there isnt a need to do it another way. Compare it with copying a audio file over USB to some devices flash which then reads and plays it. No real difference. If cables are crappy, SPDIF, USB, Ethernet hasnt any auto correction on its physics. This has to be done on some higher layers like in TCP or in an application itself. So if data-loss is happening the receiver cant handle that. The DAC will in this case just convert whats there and sound quality decrease will happen. But dont be afraid. Under normal circumstances there is no loss. Period again. So no need for some retransmiting upper-layers. Upper-layers with retransmitting functionality like TCP will just introduce jitter. Jitter is in audio a no-go. Like in VoIP audio should arrive fast and without much latency. Retransmitting is worse. As data-loss is only happening in broken setups nobody has to deal with that and like in RTP data is sent without acknowledgements on the receiver site. USB and SPDIF/AES is the same. No acking.
So under that definitions why the hell should one want to rely on the precission on arriving audio/data ticks? This isnt accurate and wouldnt work good without a buffer mechanism. Even el-cheapo VoIP hardphones have buffer to buffer voice first before playing. This is a must, because nobody can guarantee exactly arriving audio data.
So discussions about asynchronous USB (which is nothing more than a layer which controls and guarantees the data like in TCP) is totally nonsense. Because its not needed. The receiver has even in that case to buff for the best and to clock again.
So to get to your question: Sorry, but I dont know your hardware, what it does and what not. Normally I first buy something which I do know and know what it does and what it does not. But maybe its possible for you, maybe with my/our help, to understand your setup more and answer it for yourself.
Thank you Christian for trying to sort this out. I'm not sure I follow you (and no doubt that's my lack of expertise, rather than your explanation). What I glean from your comment is an opinion that an external clock is neither necessary nor beneficial because my DAC has a RAM buffer that should resolve problems that might result from jitter. Do I follow you?
Christian - You speak in very absolute terms about only one way to do things. In my experience people who speak this way do not understand the other methods of doing things. Some incredibly brilliant audio engineers who've been doing this for decades would disagree with you on much of what your saying. If there was only one way to do things, everyone would do it that way.
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