Every year we find with computer systems as with so many other products it seems that the is always something bigger, better and faster becoming available. The question is how do we validate those claims and work out which solution will fit which user whilst offering the best performance at any given price point?
Here in Scan we use a number of different tests and where gamers concern themselves with performance indicators like 3DMark and video people concentrate on Cinebench for audio the stand out test used by retailers and reviewers alike is DAWBench for audio computer system benchmarking. DAWBench’s working methodology is a rather large subject in itself and something we will be covering in later articles in much depth but here we can give a quick overview covering how it relates to audio computer system performance.
The DAWBench tests revolve around running as many instances of a given effect or audio source as possible until the CPU overloads and audio corruption is generated in the signal path. The most common variation of this test is the RXC compressor test which has been in use now for a number of years and has plenty of results generated overtime making it ideal for us to look at how performance has grown from generation to generation of audio computer systems.
The test itself is fairly simple to carry out and can be run in a number of popular sequencers including (but not limited to) Cubase, Reaper, Sonar and Protools. The template for the test can be downloaded from the DAWBench website which consists of 4 tracks of audio parts and 40 channels of sine waves. On each of these sine wave parts 8 RXC compressors are included already set up but not yet activated and it is these you switch on one at a time in order to put the system under more and more load. Whilst testing the sine wave channels that you are working with are turned down but the accumulated compressors continue to up the load on the system and you monitor the situation by means of the looping audio tracks playing through your speakers. As you reach the point where the processing ability of the system reaches its maximum handling ability the audio you hear will start to distort and break up and it’s at this point where you have to turn off a few compressor instances taking it back to the point where the audio is clean and unbroken, which when you have the audio this point you then make a note of the total number of RXC compressor instances achieved and that is your score at the buffer setting in question.
A quick real world explanation of buffer latency for those not familiar with it is this. A low buffer setting means that your input devices can communicate quickly with the CPU inside of the audio computer system and the data can be processed quickly and for real time interaction this is crucial. Something you can try yourself is setting the buffer latency in your sound card control panel firstly to it’s lowest figure normally around the 32/48/64 level and playing a note on your midi controller which you will find is very responsive at these settings. If however you raise the latency settings up to around the 1024 level or higher and now trigger your midi controller you’ll notice a definite amount of lag between the key press and the sound coming out of the speakers.
So why would we want to run an interface at 1024 or higher settings?
As you bring down the buffer figure to improve response times your placing more and more load upon the CPU as a smaller buffer is forced to talk to the CPU more often which means more wasted cycles as it switches from other jobs to accommodate the data being processed. Whilst an artist performing or recording in real time will want the very lowest settings to enable the fastest fold back of audio to enable them to perform their best, a mix engineer may wish to run with these buffers set far higher to free up plenty more CPU headroom to enable high quality inline processing VSTi’s the performance to carry out their tasks without overloading the processor which as we’ve seen before would cause poor results in the final mixdown.
Too keep the playing field level the results below have been tested with Windows 7 64bit and in all these tests we have used a firewire M-audio Profire 1814 interface to ensure the results are not skewed by using various interfaces with different driver solutions. The are better cards that will give better results at super low latencies, with the RME range for instance going down to buffer settings of 48 on the USB/Firewire solutions and even 32 on the internal models. The M-Audio unit however has great drivers for the price point and we feel that giving fair figures using an interface at an accessible pricepoint gives a fair reflection of performance available to the average user and those who are in the position to invest in more premium units should find themselves with additional performance gains. We will be comparing various interfaces in the future here on the blog and the are benchmarks being produced in the DAWBench forums which also good further reading for those of you looking for new card solutions in the meantime.
So what does the chart above show us?
The are a number of audio computer systems being tested on there from over the last few years and it shows the continued growth of performance as newer hardware has been released. The stock i7 2600 proved to be a great performer when stacked up against the previous high end Intel systems even coming close to the hexcore flagship chips from that generation. What we also see is that once you take a 2600k and overclock it as we do here the performance available is greater than the 990x for a great deal less cost wise although it has to be noted that the X58 platform has more available bandwidth which can help increase performance in some real world instances where the user is working with vast sample libraries, the results we see here are a good indicator of how the machines will run for a more typical user.
Also worth noting in the performance results above is the i5 2500 result as we use it in our entry level value systems currently. The performance is roughly half of the overclocked 2600k system and in real world terms the cost of the system is roughly half as well meaning that whilst neither unit offers better value for money than the other in the cost vs performance stakes, in instances where your recording requirements are not quite as great the value spec still offers plenty of power to get you going and achieve completion on smaller projects even if it doesn’t offer the additional cooling and silencing features we have as standard on the high end solutions. It’s also worth noting that the i5 2500 scores close to the last generation i7 930 which shows how much performance improved between the last generation and the current one.
Our high end laptop solution in all but the very lowest latency situations also proves to be pretty much on par with the last X58 based i7 930 processor which itself still offers enough power to the user to get the job done in all but the most demanding situations which means that the age of the full desktop replacement laptop is very much with us making it as easy to edit, mix and produce fully formed mixes on the road as it is to perform every night with the very same units.
Hopefully that helps explain how we rate audio computer systems in house for performance testing and will help you decide upon your own next system. We run these tests on each new range we release so keep an eye out for further articles showing testing results as new hardware reaches the market.
2 thoughts on “Audio Computer System Benchmarking”
thanks for benchmarks. concerning stock speed (on 2600) – turbo was enabled or not?
Turbo was indeed enabled for these tests at default levels, so for the 2600 cpu it would have been running at 3.9Ghz under turbo.
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