One of the toughest decisions to make when purchasing a computer for digital audio production is which processor will be best for your configuration. These days, the model numbers that designate the processors have very little to do with how fast the CPU actually is. Furthermore, the commonly used denominator for speed GHz means less than it ever has due to variables such as total number of cores, hyper-threading and turbo-boost support, hyper transports, and on and on…

As such, benchmarking digital audio workstation computers for audio production is not an easy task. There are many synthetic benchmarks available that help compare processors but these are generally geared towards gaming, 3-D animation and general productivity. What sets audio production apart from other computing tasks is that everything has to happen in virtual real time – low latencies mean that the CPU has less time to make a calculation, sometimes resulting in glitches and artifacts during recording and playback. Most synthetic benchmarks do not address audio glitches, instead they focus on the raw number of calculations a processor can accomplish in a given time. This is why we use a different, more hands-on approach to benchmark our systems.

Theoretically, a method to gauge the yield of a processor is to count how many plugins it can handle at different latencies. However, there are many variables that come into play. Different types of plugins are able to take advantage of different CPU technologies and many plugins will use up varying amounts of CPU depending on the content of the audio sent through it. Then there are VSTi’s (Virtual Instruments), which use giant hyper-sampled libraries and are very RAM dependent. To make things even more complicated different audio interfaces will give very different results.

Our goal was to simplify all of this. We chose not to benchmark a wide variety of plugins, that would be too complicated and time consuming. Instead, we focused on one plugin loaded many times. We also limited our benchmarks to one interface – the Tascam FW-1884.

Our test interface and benchmark method:

Operating system and system configuration:

All test results were measured on a systems running our specially optimized installations of Windows 7. We perform our benchmark after the computer is fully optimized for audio production. This includes BIOS and driver optimizations, removal of bloatware and unnecessary system services and other various tweaks to the system. All systems had a minimum of 4GB of RAM – however this is rather inconsequential as this is a CPU benchmark.

Interface: Tascam FW1884

We chose the FW-1884 for two reasons:

1. Since we started Reyniers Audio in 2007 we have used the FW-1884 to test all our workstations. Through experience we have found that if it works with the Tascam FW-1884 it usually works with everything else, as this interface is extremely picky when it comes to which Firewire chipset you use. Obviously we have tested our system with many different interfaces but we use the FW-1884 as a control, to test all our computers against one standard.
2. The FW-1884 provides very low round-trip latency. Many interfaces have a built-in safety buffer that will affect the true round-trip latency of your computer audio setup, from input to output via an ASIO driver. To test round-trip latency we use CEntrance ASIO Latency Test Utility – this measures the true round-trip latency and is an excellent tool to determine the true speed of an audio interface (don’t trust your host’s latency numbers – they are far from accurate). The number of plugins a project can utilize at low latency greatly depends on the buffer size you select in your interface ASIO settings. The ideal interface will allow a low buffer setting with a low round-trip latency and high plugin counts. The issue of round-trip latency is worthy of another blog post (coming soon) but below is a quick chart to show that different interfaces can have very different round-trip latency results. The lower the latency the more instantaneous audio will feel. As you can see, the Tascam FW-1884 scores very well, and because of this we think this an excellent interface to test with.

Benchmark method

As mentioned before, audio benchmarks can be rather complicated given the number of variables involved. Luckily, the guys at www.dawbench.com have come up with several benchmarks to test a system with high track counts, extreme DSP loads and very CPU-intensive workloads. Although they offer benchmarks using many different DAW hosts and plugins we have isolated our test to one in particular: the RXC benchmark. This test uses the 32-bit version of Reaper, convenient for us as we already install it on all our workstations. We use this version of the benchmark for two reasons: 1) we love Reaper and use it daily; 2) it gives us a constant platform on which to run our tests and benchmarks.

The RXC benchmark is a Reaper project that consists of:

• 40 tracks of sine waves
• 4 stereo tracks of audio content
• 320 ReaXcomp instances – 8 per sine wave

If you are curious how your DAW computer stacks up feel free to run the benchmark yourself. I have posted our slightly modified version of the DAWbench.com benchmark here.

Here’s how to run the benchmark:

1. Install Reaper 32-bit (Although Cockos updates Reaper very frequently which could introduce slight variations in your results between versions, since we started testing using Reaper we’ve experienced less than a 3% variance between different Reaper versions)
2. Extract the DAWBench-Reyniers_Audio-RXC.rar file to a location of your choice.
3. Inside the extracted directory you will find a file named “reaxcomp-standalone.dll” – place this file in your VSTPlugins directory and make sure Reaper is able to find the plugin.
4. Complete a fresh system boot
5. Set the ASIO Driver setting to one of the following buffer settings: 32, 64, 128 and 256
6. Open the project titled: “DAWbench-Reyniers Audio-RXC.rpp”.
7. You’ll notice there are 44 tracks. Tracks 1-40 have since waves and are muted by the faders. Tracks 41-44 have the actual audio content – these tracks have no effects and will stay untouched throughout the process.
8. Hit play and begin activating the ReaXcomp compressors one by one, track by track. You’ll notice the project is setup to loop continually. It is important the activated effects are spread evenly across the tracks so instead of turning on all the effects on one track, work across the project to ensure the workload is divided across all tracks. An easy way to toggle activation of plugins in Reaper is to hold the “Shift” key and clicking the ReaXcomp slot in the Track FX panels in the mixer.
9. While playing back the project, continue turning on the ReaXcomp compressors until your audio stream gets interrupted by clicks and pops. Often times the clicks and pops will only begin after the project has looped a few times so once your CPU load is in the 80-90% range it is best to slow down the rate at which you turn on ReaXcomp instances to once per loop.
10. Once a maximum has been reached, we take note of the number of plugins we were able to activate and repeat the process for each buffer size.

The Results:

By consistently running these benchmarks and collecting the resulting data for several years we were able to distill some very useful information that we now use in creating and optimizing our Digital Audio Workstation computers.

The results you see in our benchmark dictate the steps we take to optimize our workstations computers. In some systems we’ve seen an increase in performance by more than 50% after our optimizations. Through these benchmarks we have grown our understanding of how minor changes to BIOS settings, operating system and hardware drivers can make a huge difference in audio production capabilities. As a result, every system that leaves our workbench goes through a painstaking process of optimization to ensure it falls in line with results achieved with similar hardware. This not only helps us ensure our systems adhere to the highest standards, it also helps us guide our clients into a system that has both their budget and speed requirements in mind.

Some notes about the benchmark results:

• Currently, the Intel Sandy Bridge processors provide the best “bang for the buck” performance – especially the 2600k processor.
• The new AMD Buldozer (Zambezi) FX processors offer somewhat disappointing results compared to Intel’s Sandy Bridge line-up. It is however a less expensive platform which ultimately offers enough processing power for most mid-size projects.
• Intel Nehalem 6-Core processors yield the highest plugin counts on a single CPU workstation.
• Intel Dual Xeon systems featuring 12 CPU Cores (2×6) yield the highest plugins counts available to date.

It is hard to say which processor is best for everyone – again there are too many variables to make it that cut and dry. If you are a singer/songwriter and you mostly create projects by recording them one track at a time, a system based around the new AMD Zambezi processors or the Sandy Bridge i5 processor will usually be sufficient for projects requiring between 20 and 40 tracks. If you find yourself using lots of Virtual Instruments or track large bands with track counts over 40 you will undoubtedly benefit from extra CPU power, especially when monitoring directly from your DAW host software. For this we recommend the Sandy Bridge 2600k or the Nehalem i7 980 processor. For those composing symphonic movie scores using EastWest libraries or Vienna Orchestra libraries it is still the best to go with a dual Xeon setup – especially the 12 core configurations.

Indexing:

Searching:

Some of our studio projects:

Middle Rhythm Session – Les Beat (Leit Motif):

10th Concession – EP:

Kelli Debbs – Dreamers and Insomniacs:

Wolf in the Fold – EP:

What to expect from Reyniers Audio in 2011

Shipping to Canada (and other countries outside of the USA)

Recording Laptops

Season’s Greeting

The 32-bit problem

An example of a hyper-sampled instrument library is EastWest’s Symphonic Orchestra. Click the screenshot to hear it in action.

The 64-bit Solution