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- Hard Drive Buffer Size
- Auto Tune Buffer Size Windows 10
- Auto Tune Buffer Size Definition
- DAWs >Studio One
Mar 15, 2012 In PostgreSQL 9.1, prompted by some good advice from Greg Smith (blog, book), we changed the value to auto-tune to 3% of sharedbuffers up to a maximum of 16MB, the size of a single. This is all lovely, but the Device Block Size buffer is also used for playback of existing tracks and plug-in processing, and, in that context, larger buffers are better because they ease the processor’s workload. My typical buffer size for mixing would be 512 or 1024 samples.
Oct 08, 2008 Receive Buffer Size Sets the size of the socket receive buffer (SORCVBUF). This value directly affects the TCP window size. Default: 128 KB. Auto Tune Send Buffer Size Windows 7: If CTCP 6 is not enabled the user defined socket buffer size is used. If CTCP is enabled the Windows TCP stack automatically adjusts the sender's congestion window. May 17, 2019 The graphical mode of Auto-Tune Evo VST can meticulously adjust wave-forms and zooming in and out can provide minuscule corrections. In terms of options and program preferences, different options can be tuned such as the buffer size, number of undo actions and the window size.
Screen 1: The Audio Device preferences pane. A nice, short, 32-sample buffer is in use here, yielding perfectly workable latencies of less than a millisecond.
With Studio One’s advanced features, latency need not be a problem.
When audio data is moved around, it absolutely must be received and played on time, or bad things happen. In DAWs, this stability is ensured by the use of a buffer, a short-term ‘staging area’ your processor can use like counter space in a kitchen to make processes more efficient. But this buffer turns out to sit at the centercentre of competing priorities, a fact that has engendered no small amount of confusion around buffer size settings.
The Fundamental Problem
Recording is impacted by buffer size because buffering necessarily introduces latency (delay) in monitoring the source, and monitoring delay is difficult to stomach when recording a performance. It also complicates the use of virtual instruments, which get delayed as well. Small buffer sizes create less delay, so recording is best done using the smallest workable buffer size, but smaller buffer sizes also push your processor harder, and this proves to be the limiting factor in how small a buffer you can use.
In Studio One, the Audio Setup / Audio Device / Device Block Size setting in the Preferences dialogue sets the basic buffer size. For the lowest monitoring latency, set it as small as you can get it without incurring dropouts, glitches or clicks. I usually use 32 samples, or sometimes 64 samples (for high-res, high-track-count situations) when recording. The input and output latencies in milliseconds produced by the current setting are displayed at the bottom of the dialogue. Round-trip delay time is the sum of the input and output latencies.
This is all lovely, but the Device Block Size buffer is also used for playback of existing tracks and plug-in processing, and, in that context, larger buffers are better because they ease the processor’s workload. My typical buffer size for mixing would be 512 or 1024 samples. Plainly, buffer needs for monitoring and track playback are in fundamental conflict with each other.
Two Solutions
The first approach devised to resolve this conundrum was simply to reset the buffer size smaller when recording and larger when mixing. If you are using a modest or older interface that offers little in the way of bells and whistles or speed, this is the method you will continue to use. It’s a little clunky, but effective enough to have worked successfully for years.
As DSP chips became cheaper, the next solution for the buffer-size paradox emerged as dedicated DSP chips for monitor mixing embedded in interfaces. Real-time, hardware-based mixing has near-zero latency, so inputs are heard without delay and Device Block Size can be optimised for track playback.
PreSonus’s Studio 192 interface offers onboard DSP and, as well, integrates directly with Studio One, a significant feature for recording workflow. Studio One recognises PreSonus interfaces with onboard DSP or, running under Windows, any interface conforming to the ASIO 2.0 DM spec. Then it makes low-latency hardware-based monitoring available, a condition it indicates in two ways on the Audio Setup / Processing pane of the Preferences dialogue. First, the ‘Z’ to the right of the Audio Roundtrip monitoring latency will be coloured blue if hardware monitoring is recognised, and, second, only when suitable interface DSP is seen will the “Use native low-latency monitoring instead of hardware monitoring” option not be greyed out.
Screen 2: With a 512-sample buffer set for mixing, latency is too high to monitor while recording without either hardware monitoring or native low-latency monitoring. If you needed to add an overdub while mixing a session with lots of tracks and plug-ins, making the buffer smaller could result in glitching.
This last bit can seem confusing at first blush. What’s going on is that if no interface DSP is found, there can be no option to switch between hardware and software monitoring, so that option is greyed out. An interface with DSP opens the possibility of using either kind of monitoring, or even switching between them.
To use hardware monitoring, untick the “Use native low-latency monitoring instead of hardware monitoring” box. You can freely switch back and forth between hardware monitoring and software monitoring, as circumstances may suggest, just by clicking on the ‘Z’ in the Processor pane. Hardware monitoring is available only on outputs designated in the Audio I/O Setup pane as cue mixes, appearing in the form of a ‘Z’ at the bottom of the channel strip in the mixer. When this ‘Z’ is hollow, low-latency monitoring is not activated and latency is determined solely by Device Block Size. With interface DSP, clicking on the ‘Z’ at the bottom of an output channel strip will turn it blue, enabling hardware-based monitoring.
This gives us pretty-darn-close-to-zero latency, plus whatever EQ, dynamics and other processing the interface provides from its DSP. It also means that input signals are no longer run through Studio One on their way to monitoring, so you can’t monitor through plug-ins on the record channel in Studio One’s mixer. You can insert plug-ins on the input channel itself in Studio One, but the signal will be recorded with these effects ‘baked in’, even though you will not hear them while recording, since you are monitoring through the hardware, which precedes the processing,
Screen 3: When using hardware monitoring, latency on the input is of no concern. Here, an EQ is inserted on the input channel, committing that EQ to the recording. When the mixer is in the integrated window, double-clicking the meter on the input channel makes inserts visible.
One interesting technique to try if you have a PreSonus device with onboard DSP is linked hardware/software Fat Channels. Implement a Fat Channel on an interface mixer channel, implement a Fat Channel on the corresponding Studio One mixer channel, and then click the Link button. Now, the Studio One Fat Channel follows changes to controls on the interface Fat Channel. You get Fat Channel processing on your monitors while recording, and then those settings get transferred to the Fat Channel in Studio One so you can work with them after you finish recording.
I have MOTU AVB interfaces with powerful onboard mixing and processing, but they don’t communicate directly with Studio One, so I will never see a blue ‘Z’ for my interfaces. I still use the hardware-based mixing, but I have to operate it from the interface’s control software. Building a cue mix (as opposed to monitoring the main mix) in the interface control software is not much more work than doing it in Studio One. But monitoring while overdubbing is tricky on my system, and would be far more elegant with integrated control of hardware monitoring.
Even without monitoring through effects on the record channel, you still can route a send from the record channel to an effect, generally a reverb or delay, instantiated on a bus or FX channel. This effect return can then be routed to one or more headphone mixes: it will be subject to latency created by the Process Block Size, but for reverbs, that often isn’t a problem.
The Latest & Greatest
Finally, we come to Studio One’s native low-latency processing, which arrived in version 3.5. Studio One now implements a dual-buffer system, controlled by the new Dropout Protection parameter. The Dropout Protection setting determines the size of the Process Block Size buffer, which is dedicated to track playback and plug-in processing.
With two separate buffers to meet two conflicting needs, Device Block Size can be kept low for minimal monitoring latency, while Process Block Size can be set larger to accommodate playing back lots of tracks without dropouts or glitches. As with Device Block Size, estimated latency times for the Process Block Size buffer are shown in the Monitoring Latencies section at the bottom of the pane.
Screen 4: The Processing preferences pane is where Dropout Protection is configured. Note the green ‘Z’s in the Monitoring Latencies area, showing that the MOTU interfaces are fast enough to provide native low-latency monitoring.
Native low-latency monitoring requires data to be moved quickly between Studio One and the interface, so this mode only becomes available if Studio One determines that your interface is fast enough. With a qualified interface and Process Block Size set larger than Device Block Size, native low-latency monitoring is available, as indicated by the green ‘Z’ to the right of the monitoring latencies. Depending mostly on your processor, very low latencies can be achieved with native low-latency monitoring. As before, ‘Z’ monitoring is available only on cue mixes, and the ‘Z’ on the mixer channel strip should turn green when you click it.
One compromise in this mode is that although plug-ins can be used on the record channel, they must not introduce more than 3ms latency: if a plug-in is safe to use in this role, its ‘power’ button will turn green. Plug-ins that introduce more latency than this are disabled. As before, you can still send from the record channel to an effect on a bus channel, as long as you can bear the latency.
Latency also can be a downright debilitating problem when playing virtual instruments. The “Enable low-latency monitoring for instruments” tickbox puts that right, but it requires low-latency software monitoring, so ticking the box forces Studio One to use software monitoring.
Increase Buffer Size
If all this seems like a lot to absorb, well, it is. It’s all very logical, but it can take a bit of thinking to suss out the best scheme for monitoring while recording and overdubbing, especially given that overdubs can easily happen when one is far down the road mixing. A thought experiment in which you walk through the session and figure out what monitoring is needed at each step can be effective (make notes!), or you can try it in a session (preferably not on a paying client’s time!) and figure it out in action. One thing is clear: low-latency monitoring is the preferable option whenever it is available, whether in software or hardware, and Studio One offers several ways to achieve it. Happy monitoring, everybody!
One of the best features of TDM-based Pro Tools systems is the negligible latency, or input-to-output delay, while recording. So what's the best way to deal with this problem if you have only an LE system?
When working on music projects I normally use my main Pro Tools system, which is an HD2 Accel system with a 192 I/O interface. However, I also do a lot of work on LE systems as these are what many of my clients have. Often, these are used for broadcast production, and latency issues don't usually rear their ugly heads when using Pro Tools in this way, but recently I was asked by one of my clients to fly out to Northern Ireland to track and overdub some music tracks in a studio overlooking Carlingford Lough near Newry. With us flying out, taking my HD system was not a practical solution, but I have an 002R with a Focusrite Octopre in a 3U soft case, and together with some other bits and pieces loaded into my suitcase, I just got it within the 32kg single item limit.
With the 002 and 002R, Pro Tools LE offers a special Low Latency Monitoring mode.
I knew that there were some latency issues with LE systems, and sure enough, when we came to the first overdub there were problems. The singer was having some tuning difficulties, so I put Pro Tools into Low Latency Monitoring mode — and hey presto, the vocalist could sing in tune again. I was feeding the performer's headphones from an aux buss and hadn't noticed that in Low Latency mode, Pro Tools mutes the aux sends of any track in Record; I didn't become aware of this until later on in the session, when another performer complained they couldn't hear themselves. So I looked into what was going on and workarounds to keep the session going smoothly, and I thought I would share the results of my investigations with you this month.
Latency And Why It Happens
Pro Tool LE uses the processor in the computer for all audio processing, playback and recording, and to make it work reliably, audio data needs to be buffered on the way in and the way out, imposing a small amount of audio delay, or latency, in the system. The amount of latency is related to the H/W Buffer Size: the larger the buffer size, the longer the delay.
The 002 and 002R offer buffer sizes down to 64 samples.With the 002 and 002R interfaces, however, Pro Tools LE offers a Low Latency mode. This can be found at the bottom of the Options menu on Pro Tools 7 LE. When Low Latency mode is enabled, it will only work on tracks that have an input routed direct from an interface input, and not for tracks routed via an aux track, for example. If you do a Bounce to Disk whilst Low Latency mode is enabled then any aux and Instrument tracks will be ignored and so won't feature in that bounce. All plug-ins on any record-enabled track are bypassed in Low Latency mode, and any record-enabled tracks will not register on the master meters. Only analogue outputs 1/2 are available in low-latency monitoring mode, which is why Pro Tools muted my headphone feed on the session I was describing. Also, it doesn't work via the digital outputs, so you can't monitor via the digital outputs while recording in the low-latency monitoring mode.
The M Box and M Box 2, meanwhile, enable you to monitor the input signals directly whilst recording, so you can hear them without any latency. The Mix knob on the front of the M Box or M Box 2 enables you to adjust the balance of direct input signal to playback signals from Pro Tools. You'll need to mute the tracks you're recording on in Pro Tools, otherwise you'll hear both the direct signal and the delayed signal as recorded into Pro Tools.
Book Review: Pro Tools Surround Sound Mixing
Rich Tozzoli's book is an excellent handbook for anyone wanting work in surround with Pro Tools, whether for music, broadcast or film, and is full of pictures, screenshots and practical examples of real projects to help you to get stuck in very quickly.
It begins with a brief overview of how we have got to today's range of surround formats, starting with Walt Disney's Fantasia from back in 1938! Rich goes through the requirements for a surround monitoring system, including speaker placement, the ITU standard, calibration and bass management, and then looks at the best way to record for a surround project. He gives practical outlines and examples using both traditional mics and more specialised ones like the Soundfield and Holophone mic systems.
He then shows how to prepare a Pro Tools Session to mix in surround, including setting up surround paths using the I/O Setup window, routing to the interface outputs, the different ways of surround panning with either Digidesign plug-ins or the Waves 360 plug-ins, and the difference between the sub and LFE channels. He also looks at how the different control surfaces available, both Digidesign and third-party, work in a surround facility, and outlines different multi-channel mixing concepts using case studies, explaining how to use the Centre and LFE channels, and how to work 'to picture'. The final case study in this section is a look at how a DVD is designed, and explains the 'data rate and bit budget' calculations that go into the design and authoring of a DVD.
Rich covers in detail a broad range of surround-capable plug-ins including the Waves 360 Bundle, Digidesign's Revibe and Sony's Oxford Dynamics, as well as software available for surround encoding and external hardware processors like the Lexicon 960L and the TC Electronics M6000 units. He goes through the current range of surround delivery formats like DVD-Video, SACD and DVD-Audio, the Dolby range from Pro Logic to Dolby EX and the DTS system, before taking a look at an example of how our wonderful surround mixes are heard at home, albeit with a top-end consumer receiver. Chapter 11, the final chapter, looks at other applications for surround like computer games and commercials.
The DVD that comes with the book includes a number of examples, of which the first 11 are short clips showing extracts and elements of surround mixes, while the last three are complete mixes. The DVD will play on any surround receiver with a Dolby AC3 decoder, and with its accompanying written notes, is much more than an afterthought. The examples are an excellent resource, which reinforce the very practical tone that Rich takes through this entire book. I would recommend this book to everyone who works, or plans to work in surround audio projects.
Auto Tune Buffer Size Set
Pro Tools Surround Sound Mixing by Rich Tozzoli (ISBN 087930832X) is published by Backbeat Books at £24.95.
The Workarounds
Now that I've outlined the limitations of the zero and low-latency modes for Pro Tools LE, depending on which interface you are using, let's look at a range of workarounds you can use so you can run smooth sessions with happy artists.
- Reduce the H/W Buffer Size.
This works without having to resort to using the zero or low-latency monitoring modes and so should be the first choice for a solution. Go to Playback Engine menu item in Setups and adjust the buffer size as low as possible; 128 samples is a good compromise, but you will need a very fast computer and hard drive for this. The low limit for the options for hardware buffer size are set by which hardware interface you have. The 002 and 002R can work down to 64 samples (on my Apple Powerbook G4 1.67GHz machine with my 002R running Pro Tools 7.0cs5 I was able to run at 64 samples on a fair-sized Session) and the M Box 2 will go down to 128 samples, but the original M Box's lower limit is only 256 samples, which for most situations won't be enough. This has to do with the performance of the USB buss, so for low buffer sizes to work well you need a fast computer and a Firewire interface. However if you don't have either or both of these don't despair, as there are other workarounds to go at.When you're using the 002/002R Low Latency recording mode, plug-ins and aux sends on the tracks you're recording are disabled.
- Use the Low Latency monitoring mode (002 and 002R only).
This is much faster than even the 64-sample buffer size, but the down sides are that you cannot use any outputs other than analogue outputs 1/2, so headphone feeds fed via an aux buss from say outputs 7/8 aren't possible. Also, remember that plug-ins on tracks you're monitoring and recording will be bypassed as well. However, if you can work with the headphones having the same mix as your control-room monitors and don't mind bypassing plug-ins on the track you are recording on, the latency is seriously low in this mode.
Hard Drive Buffer Size
- Use the Zero Latency monitoring mode (M Box and M Box 2 only).
This eliminates latency altogether by routing the input signals direct to the outputs in the analogue domain, so short-circuiting the path via Pro Tools. However, you need to note that you will hear both the input direct and the signal coming back from Pro Tools in this mode, unless you mute any track you are recording on to. One snag with this is that when you come to try to overdub sections, the artist will need to hear what they have already laid down on the track. The workaround for this is to mute the record track at the drop-in point, but you will have to do this manually, as mute automation is suspended when you record-enable a track.
- Use an analogue mixer for headphone monitoring.
Taking the M Box's zero-latency concept somewhat further, you can use a separate mixer to handle monitoring. If you're using stand-alone mic preamps, these can be split to both the Pro Tools interface and the headphone monitor, or if you're using the mixer's mic preamps, you can put Pro Tools 'in line' like we used to do with tape-based multitrack machines. This workaround is nowhere as portable as any of the others but does give you a flexible zero-latency solution.
Auto Tune Buffer Size Windows 10
Conclusion
Auto Tune Buffer Size Definition
On balance, reducing the H/W Buffer Size is the best way of getting around the latency problem, as you can do proper drop-ins, and still have the plug-ins and aux sends active on your record tracks, so the musicians can hear reverbs and so on whilst tracking. You do need to keep your track counts down and keep the use of plug-ins to a minimum, but it is the best way to work providing you have a fast computer and fast drives. The other workarounds work up to a point, but the process of doing drop-ins is much harder as the artist will not be able to hear what they have already laid down unless you are very adept in the use of the mute buttons. So if latency is a recurring annoyance then it may be time to upgrade your computer to something a lot faster.