Optimize Pro Tools: The Playback Engine

| Mixing, Optimize Pro Tools, Pro Tools, Tips

Anytime you open up a Pro Tools session you need to ask yourself this question: Am I going to be recording, or editing/mixing? The answer to this question will determine how you adjust some crucial settings in the software to optimize your Pro Tools system. Let’s take a brief look today at tweaking the Playback Engine in order to get the most out of your computer while recording.

Reducing Latency

If you are using Pro Tools to record for example (perhaps you’re sitting down to take the One Song One Month Challenge and you want to lay down your main guitar parts) then you want as little latency as possible. What is latency you ask? It’s the time it takes for your audio to be converted to digital information (in your audio interface), run through your software, then turned back to an analog signal coming back out of your audio interface. The result…an annoying echo or delay effect that throws off your timing.

Setup1Some interfaces have a workaround for this on the actual unit itself (the Mbox 2 Mini for example uses the Mix knob to counteract this). But one thing you can do in all Pro Tools systems is to reduce the Hardware Buffer Size. To do this, simply navigate to the Setup menu and choose Playback Engine.

If you have something like 1024 samples in the H/W Buffer Buffer Size drop down, then click on it and choose the lowest sample number available to you. It might be 128 or maybe even down to 32. Setup2Choosing a lower buffer size will noticeably reduce your audible latency and make recording much better. On the flip side if you need to edit or mix, you’ll probably have error messages pop up with such a low buffer size. Simply change it to the highest option available (1024 is good).

Your House In Order

While you’re at the Playback Engine settings you should also confirm that your RTAS Processors drop down is set to the appropriate option (this depends on your computers CPU of course). Setup3And as far as the CPU Usage option, I recommend you don’t go much higher than 85% as you need your CPU to also efficiently run your operating system in the background as you are recording in Pro Tools.

Nice And Easy

Voila! You have quickly optimized Pro Tools using the Playback Engine settings. Little tweaks like this go a long way in ensuring a smooth and enjoyable Pro Tools session. Enjoy!

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36 Responses to “Optimize Pro Tools: The Playback Engine”

  1. Keilan

    Thanks for this, dude. I was trying to lay down some tracks today and I kept getting that annoying echo sound that made it difficult to record. But I had forgotten that while mixing last time I had upped the h/w buffer size. Thanks for saving the day again, Graham!

    Reply
  2. Junior

    The problem i have, I have a focusrite saffire pro 6, pro tools 9, audio techa 2050 mic, my pc is a ixtreme, i5 core, 11gb ram, 750gb hard drive.
    if i put the h/w buffer below 1024, the track won’t play, its like dot dot dot dot dot… so i need to put the buffer up to 1024, but when i do, i get this huge delay in the headphones? any help?
    Cheers

    Reply
    • Graham

      Hmm…check that you have the latest Focusrite drivers and version of Pro Tools 9 installed. Then use the Mix Control software to monitor your AT mic. Mute the actual track in pro tools and just listen to the no latency version.

      Reply
  3. Junior

    Hi, I’ve checked for the latest updated drivers, i have them, but the plugin suite for my saffire pro 6, the mix control software won’t seem to open up in pro tools or separately? Installed and uninstalled a few times? plus i dont really know where to find it too? is it where autotune, reverb etc.. etc.. would be?
    I can record with a mute track but i just prefer hearing my vocals too….

    Reply
  4. Graham

    Sounds like you should email support at Focusrite. (They are pretty responsive!). What you need is to be able to open the Mix Control software, as this allows you to monitor your tracks with super low latency. It should be a separate piece of software that you open up before Pro Tools.

    Reply
  5. Junior

    I’ve got a panel that says “focusrite usb asio control panel.
    Its got one slider that says
    Buffer length 11ms and then size in samples 512 ? is that it?
    Cheers. thanks for this.

    Reply
  6. Junior

    I’ve got a panel that says “focusrite usb asio control panel.
    Its got one slider that says
    Buffr length 11ms and then size in sample 512 ? is that it?
    Cheers. thanks for this.

    Reply
  7. Graham

    Sounds like that might be it. I’d definitely check out the Focusrite website and knowledgebase for more details on the software you have.

    Reply
  8. Junior

    Check the website, not much help really. They said that panel is for the driver settings, but i should manually change them in pro tools but when i bring the buffer size anywhere near 128 or 256, it won’t play.

    Reply
  9. Brandon Michael

    what sample rate and bit depth should i usually set a session up with? I am using a dell XPS laptop with windows 7 and an MBOX 2. Also, when I bounce a track for a stereo mixdown what should i set up the sample rate and bit depth as, same as the session?!?

    Reply
  10. Brandon Michael

    thank you so much! such a big help! Also, one more thing…. how can i normalize the volumes of my tracks? For instance.. I can’t mix some tracks down at the same level as others. It just distorts out.. i know there’s a way to throw gain on an MP3 or WAV without compromising the quality of the mix. how should i go about doing this!?

    Reply
  11. Brandon Michael

    Audio normalization is the process of increasing (or decreasing) the amplitude of an entire audio signal so that the resulting peak amplitude matches a desired target. Typically, normalization increases the amplitude of the audio waveform to the maximum level that does not introduce any new distortion other than that of requantization.
    Normalization is often used when remastering audio tapes for CD production[citation needed], in order to maximize the signal level while not changing the signal to noise ratio. It is often combined with dynamic range compression and hard limiting to increase the apparent volume of a CD. It is typically applied along with other audio and digital processing, such as dithering.
    Normalization is commonly amongst the functions provided by a Digital audio workstation.

    Replay Mp3 Gain works by first performing a psychoacoustic analysis scan of the entire audio file to measure the perceived loudness and peak levels. The difference between the loudness and the target loudness is calculated; this is the gain value. Typically, the gain value and the peak value are then stored in the audio file as metadata, allowing Replay Gain-compliant audio players to automatically attenuate (or in some cases amplify) the output so that such files will play back at similar loudness to one another. This avoids the common problem of having to manually adjust volume levels when playing audio files from different albums that have been mastered at different levels. With lossy files, another benefit of Replay Gain scanning is that the peak information can also be used to prevent loud songs from clipping.
    Replay Gain implementations usually involve adding metadata to the audio without altering the original audio data. MP3 files usually use ID3v2 or APEv2 tags. CD players and other legacy audio players do not support Replay Gain metadata. Nevertheless, some lossy audio formats, such as MP3, are structured in a way that they encode the volume of each compressed frame in a stream, and tools such as MP3Gain take advantage of this to change the volume of all frames in a stream, in a reversible way, without adding noise.
    ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
    I found this on a website.
    HOW CAN I UTILIZE THIS TO HELP ME!!!!

    Reply
  12. Graham

    This is not an issue of normalization. It’s an issue of limiting the mix with a compressor/limiter. Typically a mix is mastered before it is released and this helps with the overall gain and level of the mix. If you’re having volume issues of mixes try limiting them when you’re done.

    Reply
  13. Brandon Michael

    thanks Graham! i am doing some trial and error with the compressor/limiter to try and balance the level of all the songs I want to put on my demo CD. It seems like it’s working. Basically i mix down a song with all the stems and individual tracks into a stereo wav and then throw that into a new session to master it and use the compressor/limiter. Is this what the pro’s do with their mixes?

    Reply
  14. Brandon Michael

    Also, I see here on tunecore.com that Universal charges $70 a song in which they do the mastering for you. is that worth it to go that route when i eventually want to release an EP independently?

    Reply
  15. Hedley K

    Hi Graham, Big fan of the site. I’ve had problems with H/W buffer size and CPU in the past, one question I have is with multiple processors (I have 6) how many should be selected as RTAS Processors. Does this effect the percentage that should be selected?

    thanks

    Reply
  16. Muler

    Protools 9, brand new imac i5 processor with 4gb memory, can not play back protools session with 16 tracks. Autotune, delay and reverb plug ins on 6 tracks only. Played with the highest level of buffer but doesnt help. It will not even bounce. Does increasing memory to 8GB helps?

    Also does the use of aux track for processing audio effects helps compared to each plugin on a the tracks?
    Thanks!

    Reply
    • Graham

      Muler, is this without an audio interface? What interface are you using? Are the tracks on an external hard drive or the system drive? What version of mac os are you using?

      Reply
  17. andrew bennett

    Hi Graham, I just came across your site while searching for info on popping noises in my final mix.
    Any thoughts on why this might be happening.
    Thanks in advance.

    Reply
  18. Bluesdog

    Hi Graham,
    I have the most powerful iMac (3.4GHz i7 16GB + OS X Lion) running PT 9 with a Liquid Saffire 56 (all drivers, etc.. up to date). One would think that with a computer as powerful as this it should be able to perform fluently at whatever settings but I still get the ‘overload’ message from time to time (in an 8 to 16 track song).
    Could you talk us through some other settings like:
    - Delay Compensation Engine
    - DAE Playback Buffer
    - Cache Size
    And other settings I might have over looked.

    Thank you Graham.

    Reply
    • Graham

      Well it all depends on how many tracks you’re running, the sample rate, and the amount/type of plugins you want to use. But generally I put the playback buffer at it’s largets when mixing. Also, tweak your settings to you’re using all but one of your cores. This leaves power for the OS to run as well.

      Reply
      • Ben

        Dear Graham,

        I always have this question:

        If I record audio track (guitar with guitar proccesor plugin on) with 32 samples buffer size I get smooth sound, no latency. If I change buffer size to 1024 when mixing does sound quality of the track drop (slight latency?)? That would mean I mix with tracks that sound slightly different from what was recorded.

        Guitar plugin is NI Guitarig 5

        Thanks a lot :) Love your site!

        Reply
        • Graham

          Ben, the buffer size only affects latency when recording in real time. Not audio quality. Your guitar tracks will sound totally the same.

          Reply
        • Brett

          He’s right. Try this. If you change the hardware buffer size from 128 to 1024 you will hear that at 1024 the sound becomes cleaner and not as thick as it was at 128. Does anyone else hear this too?

          Reply
  19. Helge

    Thanks for sharing this! My main problem right now (ProTools 10.3.3 on the new iMac 27″ late 2012 model with 16 Gbyte RAM and fast Firewire 800 disk connected) is that when I try to bounce more than twenty minutes to my hard drive, I get the error that “couldn’t read audio fast enough to continue”. I have tried different settings in ProTools, but I just can’t make it work. I only have four stereo tracks in ProTools, a few filters and one simple reverb, so it’s a mystery to my why ProTools can’t read the audio files and bounce my mix. I never have this problem with Logic and Live, they even (as we know) bounce much faster than real time – and with no error messages. Any help is very welcome! By the way, It doesn’t seem to change the performance of ProTools at all if I increase the amount of processors in the playback engine window. With my previous version of ProTools, I got the best result (most steady) when choosing only one processor! I’m confused….

    Reply
    • Graham

      Not sure what the issue is here. With the processers in Pro Tools, I usually set it to one or two fewer than I have available so that Pro Tools and the OS run smoothly.

      Reply
    • Graham

      Sounds like it’s taking a while to read your hard drive. Maybe you can defrag the drive?

      Reply
  20. Brian Vickio

    Hey Grant,

    Going to apply this when I get home. Last night during tracking (Protools 11 Native with Orion 32 interface) When my HW buffer was lower then 1024 i would get clicking and poping. No plug-ins and the ignore errors box unchecked in the playback option. I also have the session set to record in 32-bit float and my interface is factory set to 24-bit. I’m recording a 44.1 also. Wondering how to get my latency under control without the clicking and poping? Any Ideas?

    Reply
  21. will

    I hit record on protools M powered…….and there is a signifigant pause…..up to a minute then itll start recording…mac os 10.8.5 but im using m audio drivrrs for os system 10.8.4. as it seems they NEVER have the drivers i need

    Reply
  22. Ricardo Bullemore

    for low sample rates here is a formula:
    (I/O Buffer Size/Sample Rate)*2
    if your buffer size is set to 128 and tour sample rate to 48khz then:
    128/48=2.7 now 2.7 multiplied by 2 =5.4ms of latency
    now if your buffer size is set to 128 and tour sample rate to 96khz then:
    128/96=2.7 now 1.3 multiplied by 2 =2.6ms of latency
    you get less latency at higher sample rates.
    The quotient of the buffer size and sample rate are multiplied by two to account for both the input and output buffers.

    Reply

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