{ "numMessagesInTopic": 18, "nextInTime": 1914, "senderId": "O3UPgxao-XOGqUnNOBHV9dQACeVdlH_3O_Gpe5B-R4lwWfLfY0DKORY2OG5BKHrqtsXtL2zA_to88wMn1W-Ua7DLPb_O1p3Ncd-o", "systemMessage": false, "subject": "Re: [01v96] Re: This is the kind of case I'm going to be getting for my V...", "from": ""Ronny Morris" <romo1@...>", "authorName": "Ronny Morris", "msgSnippet": "... From: Carol To: 01v96@yahoogroups.com Sent: Sunday, August 08, 2004 7:45 PM Subject: [01v96] Re: This is the kind of case I m going to be getting for my", "msgId": 1913, "profile": "ronnymorris2001", "topicId": 1602, "spamInfo": { "reason": "0", "isSpam": false }, "replyTo": "LIST", "userId": 165376131, "messageBody": "
\n\n\n----- Original Message -----\nFrom: \n Carol \n\n \nSent: Sunday, August 08, 2004 7:45 \n PM\nSubject: [01v96] Re: This is the kind of \n case I'm going to be getting for my V...\n\n\n
> Also Tom, I'm running full range Community cab's for \n mains. If
you are running bi or tri amped, Doug is definitely the one to \n talk
to. Sample latency from input to output on the 01v changes from \n
sample rate to sample rate. At 44.1 and 48k, it's only about 14 \n
samples or .3ms. At 88.2 and 96k it's only about 26 samples
still \n .3ms. You could expect delay on the ADA to be close to this at
14 samp-ps, \n in most situations it's probabaly inconsequential.
> \n
>
Ronny
The time it takes for a signal to go in the V , \n be converted to
digital and turned back into analog is the units \n latency.
On Page 283 of the manual you will see Yamaha's Signal delay
( \n latency ) or time it takes for a signal to pass thru the board.
At 48 khz \n it is rated at 1.6ms. Most likly at 1 khz.
Pink noise through the V gave me \n a 1.18 ms latency impluse time. This
of course is with no eqing or channel \n etc delay. Most likly the
impluse time locked onto a higher than 1 khz \n signal. 1.6 ms is just
under 2 feet of delay time. This would be the true \n latency of the
board. .3ms is a little short. I think you are up on this \n better than
me but maybe you missed this one.\nActually you are \n way up on driver array delay stuff than I am, Doug, but I may be able to \n clear up a few things while you are still in the digitial domain. Your delay \n latency is your total throughput time. You are talking analog signal to \n analog signal, "with the digital section of the board in between". Once the \n signal is digitized at 48k, it takes .3ms, to go from backside of the \n ADC via the digital outputs and to the next digital device. Tom was \n talking about going to the ADA with the ADAT for his subs. Signal will remain \n digital until it exits the ADA with approx the same total throughput of the \n v96, with the exception of how many samples it takes the signal to go through \n the digital side of the ADA, which likely isn't more than 14 \n samps at 48k because that ADAT signal for the subs is still digitized \n until it exits the ADA. You don't need to calculate total delay \n latency from the front side of the input ADC. The signal will be \n oversampled and this will add to throughput delay or what yammy \n calls delay latency. The V's use a 128x delta-sigma modulator for ADC's and \n DAC's (64x at 96k), which consists of an analog modulator/comparator, a \n digital filter and a decimation circuit, which will return the oversampled \n signal to the internal sample rate selected. In essence your signal just \n doesn't pop in and go, it gets oversampled by returning from the LP filter to \n the comparater in a loop through the logic chip (oversampleing), clock and \n finally the decimation filter which also determines the word length, this is \n where most of the latency comes from in your 1.6ms example of total throughput \n from annie input to annie output. All of the 01v mic input \n signals will be locked sample for sample after the 128x \n oversampling A/D conversion and will exit the decimation circuit of \n each DAC at the same time, BUT those freq's will be eq'd in the \n digital domain. The internal 01v doesn't know frequencies as time constraints, \n not like freq's on analog circuits. Unlike analog, to the 01v96 the \n representation of all frequencies are just ones and zero's, while they are \n inside the board. It's at the DAC or rather on the analog side \n of the DAC where you should start your calculation for your frequency \n deviations going to highs, mids, and subs. IOW, if you are delaying array's, \n don't count throughput time from the mic pre's. Count from where each DAC \n is in relation to the freq deviation going to amps/speakers. While it's true \n that you are assigning your LPF's and HPF's internally within the board, \n the total throughput or delay latency times "minus" the 14 samples \n won't be a factor and while digital all freq's will travel at the same time. \n\nTo clearify by \n example. When you 'digitally" cascade two 01v96's at 48k, you delay the master \n V's input channels by 14 samples, because that's how long it takes the \n digitized signal from the slave to arrive to the master. Than they all output \n the master at the same time. Remember we are in the digital domain. \n\nOne thing that I \n did notice where I was incorrect about, the throughput being the \n same at 48 and 96k. Total throughput at 96k is .8ms and at 48k it's like you \n said at 1.6ms. 96k has exactly half the total throughput time. I would \n have expected it be the same, as a second is a second, no matter what sample \n rate you are running. However, my guess is that the oversampling at 96k is \n faster, because at 48k oversampling is 128x and at 96k, oversampling is at \n 64x. Exactly half the oversampling rate, exactly half the through put time. \n Another thing that threw me off is processing taxation. At 96k it's much \n slower when a processor is printing an effect or an eq change, to a \n signal than it is at 48k. This may be why you can only use two fx processors \n in 88.2 and 96k modes. But in any case, I don't see where total throughput \n time is what you need to be concerned with and to get accurate delay time \n when using digital eq's for each frequency range speaker, is to calculate at \n the DAC's output. IOW's, 140Hz will exit the DAC at the same time that 10k or \n 20k will, becuase the digital representations of these freq's (ones and \n zero's) remain time constant and sample for sample accurate until they exit \n the decimation stage at the DAC and are than turned into \n analog.\n\nYou also mentioned \n this:\n\nAlso if you are running the 100hz highpass in the midhigh boxes then \n\n
don't use the 50hz lowcut on the L/R outs. This would cause phase \n
issues even at that low of a point. I have seen two filters run in \n
series that returned a system to flat!\nIt's been while \n guys but if I remember correctly, that's because a 100hz signal is 180 degrees \n out of phase at 50Hz, you use the lowcut you null the doubled \n frequency. Where you can get into problems with this and need \n a cut filter at 50Hz, is for example getting low end rumble coming through the \n mains, like a drummers kick on a wooden stage that is going through the wood \n (signals travel faster and are less attenuated going through wood than air), \n traveling up his mic stand, into the mic and back through his monitors causing \n a loop that attenuates to a low end feedback (rumble). Everytime he hits the \n kick you hear a whhoooommm sound instead of a good clean short \n transient thump. Rather than taking it out of the mains and risk nulling \n doubled frequencies, just cut everything below 50Hz on the drummers mic or \n drop the monitor graphic slider at 50Hz all the way down on the drummers \n monitor. You'll be able to get the drummers monitor louder with out rumble \n going to the mains via his vocal mic. He'll love you for it and you'll finally \n be able to get the kick up loud enough in the mix and comparable \n to the bass guitar running amp line out.\n\nLike I say, I'm going by memory of \n what I learned years ago, so if anyone can correct me please do. \n\n\nKeep On \n Trackin'\n\n------Ronny Morris - Digitak \n Mastering------\n
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Kindest regards
Doug