iLive 2

This topic contains 52 replies, has 28 voices, and was last updated by Profile photo of dnxmirrorsounds dnxmirrorsounds 7 years, 9 months ago.

Viewing 15 posts - 16 through 30 (of 53 total)
  • Author
    Posts
  • #33420
    Profile photo of vilddyr
    vilddyr
    Participant

    the white paper is about converter latency, not processing latency. A plug-in from a sequencer will always report its latency in samples, and this does not change with samplerate. This means double samplerate, half latency. I would just suspect the same behavior from processing in a digital desk :) this would also explain why most desks out there are not phase coherent, when introducing parallel processing. More processing – more latency.

    #33421
    Profile photo of Stix
    Stix
    Participant

    Ah yes. thanks vilddyr. I had taken DSP out of the equation.
    Now – where are my shorts?

    Cheers

    Richard Howey
    Audio Dynamite Ltd
    IDR48/IDR16/T112/R72/Mixpad,Tweak,
    Dual M-Dante/DVS, 17″MBP/Logic 9/Custom Mackie Control

    #33422
    Profile photo of woutert
    woutert
    Participant

    quote:


    Originally posted by Stix

    Quote:
    Originally posted by millst

    While this white paper doesn’t specifically talk about converter latency it does misspell some myths about sample rates and audio timing.


    I’m just doing some drawing… it’s not so hard to prove that what he writes in his paper is wrong…

    And although frequencies above our hearing limit are not audible to us, they are still present in audio material and can indeed affect the timing of lower frequency wave peaks!
    For some reason, people seem to forget that analog audio can be (and is) more than sine waves!

    Actually, I can get pretty mad when I read such nonsense from people who present themselves as specialists.

    Wouter
    IDR32, R72, Dante, Mixpad
    laptop, TP-Link TL-WR1043ND

    #33426
    Profile photo of millst
    millst
    Participant

    quote:


    or some reason, people seem to forget that analog audio can be (and is) more than sine waves!


    Exactly!!!
    Which is why I trust my ears before any maths.
    Sometimes its nice to be able to understand and explain what you hear with your ears using maths and computer science, but at the end of the day, all I really have to go on is what I hear. Sometimes things in audio don’t make sense.

    I really dislike what I hear coming out of many of the other desks on the market compared to the iLive. The desks I dislike seem to be running at 96k or higher. The iLive runs at 48k.

    This tells me one of three things…

    Either everyone else is crap at making desks that sound as good as my old analogs OR it has something to do with the sample rate OR I’m nuts.

    The problems all occur when you start summing channels. The 96k+ desks sound fine with a single mic in them, but when you have 40ch of band or orchestra, it just doesn’t sound right in the 10k to 15k range. Sample rate is the only common theme I can find between the desks that sound right with lots of channels and the ones that don’t.

    There are no doubt a lot of myths, facts or otherwise.

    What I don’t want to see is A&H going down the path of 96k just because everyone else has and just because it is a marketing / sales tool (can everyone agree on that?).
    That would be the wrong reason.

    Going down the path of 96k to reduce latency by 0.2ms (random number chosen) while sacrificing audio quality would also be the wrong reason because in my opinion, the latency is already best in class. (this one is still up for debate as some people seem to think latency is an issue).

    If however, they can go down the 96k path, reduce latency AND make the desk sound better. Well that’s just fine and dandy by me and I’ll shut up and put my order in.

    This is a useful discussion, I’m not sure I’ve learned an awful lot because there seems to be so many contravening ‘facts’ from ‘professionals’ that it is very hard to get a full grasp on the issue without getting clouded by myth or pseudo science.

    I guess that makes the issue all the more important to highlight to make sure we (as the market) are not putting pressure on A&H to do something that would not actually be in our best interests.

    #33427
    Profile photo of tk2k
    tk2k
    Participant

    quote:


    Originally posted by millst

    quote:


    or some reason, people seem to forget that analog audio can be (and is) more than sine waves!


    Going down the path of 96k to reduce latency by 0.2ms (random number chosen) while sacrificing audio quality would also be the wrong reason because in my opinion, the latency is already best in class. (this one is still up for debate as some people seem to think latency is an issue).

    If however, they can go down the 96k path, reduce latency AND make the desk sound better. Well that’s just fine and dandy by me and I’ll shut up and put my order in.


    What you are saying about the 96k doesn’t make any sense from a technical perspective, in terms of the summing of inputs. Its possible a lack of phase coherency is related to the ‘problem’ you are heading, but that’s got to do with DSP architecture, not sample rate, or sample rate mixing.

    If this were true, no one would bother recording at 192k for studio work. The recorded audio world is picky enough we would have shot down higher sample rates long ago if what you are talking about was inherent to it.

    iDR-48, T-112, Mixpad
    College

    #33430
    Profile photo of millst
    millst
    Participant

    I’m not suggesting it makes any sense from a technical perspective.

    However, the under the hood architecture of studio based systems is quite different from the live scene however.
    Studio systems have a more flexible DSP approach, usually PC based while live has a dedicated DSP approach.

    I’m not sure it is an apples with apples comparison.

    Toby

    #33433
    Profile photo of tk2k
    tk2k
    Participant

    quote:


    Originally posted by millst
    However, the under the hood architecture of studio based systems is quite different from the live scene however.
    Studio systems have a more flexible DSP approach, usually PC based while live has a dedicated DSP approach.

    I’m not sure it is an apples with apples comparison.

    Toby


    I’m not sure that’s really true anymore

    HDX is entirely FPGA-DSP-based (whereas HD Native is host based) Digico SD# is all FPGA software-dsp based, Bluefin 2 (calrec) is also FGPA based, with the SC48 is somewhere between. Pro2 is FPGA based as well.
    While certainly there’s instruction sets for each application, the differences are far harder to identify.

    Yamaha is the only manufacturer with a true inflexible DSP-based approach still on their flagship (CL). Venue is also still hardware DSP based.

    Side note: Does anyone have any honest info on the iLive/GLD DSP? I don’t believe it is FPGA based, but its gotta be partly software based since we can reconfigure it so easily.

    iDR-48, T-112, Mixpad
    College

    #33435
    Profile photo of jimvoyager
    jimvoyager
    Participant

    Congratulations guys.
    This has to be the most interesting thread for ages!
    My head was spinning just reading what you guys have writen.
    I had to sit down for a minute.
    This is what forums are for!
    I notice that none of the A&H guys have dared to chip in….but I bet they
    Are watching.

    ;)
    Jim

    R72 & IDR32
    MacBook pro

    #33441
    Profile photo of jabney
    jabney
    Participant

    An RIAA curve available on an Analog input pair.

    A fifth floating band (ala UREi) on the channel EQ.

    Designer plugins (Celmo, DAS, and emulations e.g. Pultec, Lang etc.)

    best,

    john

    Strange that a harp of thousand strings should keep in tune so long

    #33442
    Profile photo of jeffs7
    jeffs7
    Participant

    quote:


    Originally posted by millst

    I really dislike what I hear coming out of many of the other desks on the market compared to the iLive. The desks I dislike seem to be running at 96k or higher. The iLive runs at 48k.


    The Yamaha M7CL and CL series run at either 44.1k or 48k and it is available as a setting on each console. I personally have only ever observed it set to 48k. The PM5D can run at 44.1k, 48k, 88.2k, or 96k. There are lots of reasons why a mixer can sound bad, from poor mic preamps to bad processing algorithms. Personally, I would suspect one of these long before I would ever suspect something inherent to the clockrate itself.

    I definitely agree with the request for faster boot up and operation and a lower noise floor. The waiting time for the surface or editor to connect is ridiculous! I have yet to buy the iPad app because am afraid to find out it has the same long wait time and then be out the money. I would also like buttons that did not feel so flimsy.

    As far as software goes, I think it would be nice to be able to save the layer you are in on each bank in scenes and to allow more special characters (!-,: etc.) in scene and show names.

    The biggest thing I would not only want, but expect, is better technical documentation about the hardware and software. In fact, I would like that about the current generation as well.

    ~Jeff

    #33459
    Profile photo of TomD
    TomD
    Participant

    I wish it run at 96 too.

    Bob Katz wrote about that in his mastering book (upsampling for better processing treatment). If I remember correctly, 96 is ok but 192 introduce other problems.

    anyway, digico can do both so A&H must offer the option too.

    Even if it was a commercial argue, if a client wish to record the show at 96, you could just say “ok let’s do it” or explain him what’s your point on aliasing.

    It would be great it group could be send into group and FX like an input (as protools with mixbus).

    Thomas
    T112 IDR32 Dante=>MBP PTHD9

    #33461
    Profile photo of mervaka
    mervaka
    Participant

    quote:


    Originally posted by tk2k
    Side note: Does anyone have any honest info on the iLive/GLD DSP? I don’t believe it is FPGA based, but its gotta be partly software based since we can reconfigure it so easily.


    The DSP in fixed racks (and probably modular racks also) are a lot of Motorola/Freescale DSP56k processors, and I think an FPGA or two to handle the internal routing. Next time my iDR48 is open I can take another look. I’ve always admired the design!

    #33469
    Profile photo of vilddyr
    vilddyr
    Participant

    Though it seems people have taken it that way, I’m not particularly interested in wether the iLive system does 96k og not. 48 is fine by me. This is certainly not where I meet the limitations of the system. My wishes would be:

    1. A better compressor. The current one is extremely flexible of course. Sidechain, int SC, Multiple algorithms, release time in dB/s, blend. All I really want is a more simple compressor, that sounds more “right”. The Vi desks have the most simple controls, and the best sounding compressor on digital desks. The iLive’s compressor is one of the reasons I find many new techs have a hard time “getting there” quickly, when they first meet the system. Simplify the controls, improve the algorithm(s). Peak and RMS is more than enough, if the basic sound is good! And give us a proper release parameter, milliseconds like everyone else.

    2. The Limiter is now useless, and completely opposite of the compressor. No control at all. It is way too slow, obvious, and at it’s best it can be used to squash a DJ from smashing everything red. I really don’t understand it’s purpose.

    3. Better feeling build quality of surfaces. They are simply too expensive for the kind of feel you get from a T surface or R72. Even the GLD is already much better, while not very good. I don’t think there’s anything wrong with the build quality as such, but the look and feel is really not impressive. If you wanna play with the big boys, this should be addressed, though some of course will think that sounds ridiculous.

    4. Lower noise floor, I really agree.

    5. MUCH better startup and loading times.

    6. an even better de-esser. Maybe the ability to apply broadband reduction, or a Q function.

    7. More FX slots, and the ability to return outboard FX to FX returns instead of input channels.

    8. Keep being totally awesome at an affordable price!! [:D]

    [8D]

    #33480
    Profile photo of Wolfgang
    Wolfgang
    Participant

    some ideas:

    – sidechain for dynamics in inputs AND busses, with flexible routing from all inputs and all busses

    – a ducking-function in busses, with sidechain and with adjustable max. depression.

    – possibility to insert channels, when a new channel must inserted in an existing channel-list (inclusive preamp-gain!).
    not long ago, I must insert an additional tom-channel – and move all 30 channels behind them, per copy and paste preamp, channel processing and mixes = 90 actions = what a stupid job, and not one mistake is allowed!
    ok, in livesituations I can simply insert a channel from anywhere to the new position on the surface. that´s easy. but for coming events, I need a correct channel-list, I´m a german ;-)

    – possibility for quick safe channel-functions in more of one scenes (for example when another bassplayer will be integrated in a musical-band)

    – a parametric-EQ section with 6 bands.
    the lowest an the highest band can used as locut/highcut/bell/shelving-filter. this EQ is for outputs – and for inputs(!) and give us a maximum of flexibility.
    why? all desks in this world have a 4-band input-EQ, whitch is in 99% functionally adequate. but, is this really necessary forever? sometimes, I need an additional filter for example for feedack-reducing in critical live-situations, or a locut, 4 band-EQ and a highcut in a channel.
    maybe, the hardware-EQ buttons can have the normally 4-band EQ for better overview, and the additional filters can be used per touchscreen.
    I think this can be an innovation in mixerdesign – and an unique selling point!

    – automatic-mixer function to integrate the iLive in fixed installations, or for better handle double microfone-installations at lectern-situations (witch is mostly uneffective, but sometimes the customer needs it for different reasons)

    – two screens for jobs with two technicans in the same time or for better overview.

    – screen with overview of all dynamic-LEVELS (gates, compressor, limiter/deesser)

    – don´t forget: the screen must have a good visibility also in open-air requirements.

    – like “vilddyr”: a better control for the limiter and deesser.
    – like “vilddyr”: a better quality for the surface, please!

    – dynamic-EQ as alternative compressor-function in input-channels.

    – Dante network inclusive gaincontrol. i think this is really the future.

    – gaincompensation for FOH + MON systems.

    – a much faster boot-time

    – much faster scene-safe and -recall (for complete scenes)

    – more FX-blocks, and with a distortion-FX! – sometimes the iLive sounds too clean ;-)

    – samplerate:
    48kHz is IMHO good enough for live-situations. I have no problems with it. the latency-time in the existing iLive is very good!
    but: why not, with 96kHz we get a latencytime like analogue desks? ;-)
    maybe you can make a switch between 48k and 96k?

    oh, this is a long list… sorry.
    and now: I hope you can understand my english ;-)

    #33539
    Profile photo of mumu
    mumu
    Participant

    full carbon surface!!!![:D][:D][:D]
    digico and all the others are still way to heavy.
    i already love it now moving the console around,
    in case there are not as many people coming as expected-go where the crowd is – besides single channel delay sone of the main advantages of
    light weight console (rember the days with the monster muco 20 external effects e.t.c.)

    cheers
    dave

    allways latest firm and software
    iLive-144/t-80/idr-10 /idr-48/dante/pl-6/eyepad 1/belkin router/

Viewing 15 posts - 16 through 30 (of 53 total)

The forum ‘Archived iLive Discussions’ is closed to new topics and replies.