"Choosing the right audiophile playback software can be a daunting task. While audible differences can occur in going from an entry-level software like iTunes to one of the audiophile playback engines mentioned below, the transition between high-end software boils down to a preference between real cherry flavor and artificial cherry flavor. ... Some people will hear a tremendous difference while others will not. Because many operating systems can be optimized outside of playback software, the benefits of these audio applications may diminish. This doesn’t mean they make no difference, it just explains why some people will hear a tremendous difference while others will not. There are lots of layers here, and I’ll talk about them more in-depth in our upcoming optimization guides. Before diving into the software comparison, I need to address bit-perfect playback. There are three camps here. ... Finally the third camp, my camp, gets two paragraphs because it's my camp and I'm writing this. Let's all start by agreeing that audio is a real-time process. Even if an application loads data into memory for processing, everything before and the whole operation after is a real time operation. Real time processes in a computer take the form of a square wave, specifically a pulse width modulation. This pulse width modulation is an analog representation of what we conceptualize as a digital signal and is created by voltage in the power supply. This PWM signal has both amplitude characteristics and timing characteristics. The timing, or duty cycle, along with the amplitude determine the frequency response of that square wave. A computer is made up of billions of transistors, all switching very quickly to changes in logic (mathematical algorithms created by the operating system and software). Based on the input voltages, logic switches create a new version, a duplicate, of the square wave (either theoretically identical or altered). That new version of the square wave is also created from power in the power supply. Because audio is real time, there is no error correction that can be done to this square wave, any resulting wave form IS your music. Looking at the concept of bit-perfect, it's arguably impossible to have bit perfect playback in a real-time system because there are no bits. If the power supply introduces noise or there is jitter on the square wave this results in a square wave that is not identical to the original. Because the square wave is an analog signal it is still susceptible to noise and distortion. A square wave, however, reacts a little differently than its sine wave counterpart. Jitter is an alteration of the duty cycle, when that jitter hits the digital interface chips, a DAC for instance, that jitter is seen as an amplitude error and creates an alteration of the frequency response. Amplitude distortion itself is created by noise voltages that either add or subtract from the amplitude of the square wave. This introduces harmonic content into the square wave that shouldn't exist in the music. The square wave may still resemble a one or a zero, but it contains additional frequency content. So as far that bits are concerned, it's bit perfect, but with additional harmonic content that shouldn't be there. So, high-end playback software works to buffer the audio signal and keep as much of the processing in the non-real-time zone (memory playback) as possible. The next step is to create as few duplications of the square wave as possible and get it to the computer's output as quickly as possible so as to avoid the introduction of jitter and amplitude errors. All of the software below is bit perfect, the camp you pitch your tent in shouldn't affect the software you wish to use, just how you choose to integrate it into your system." http://www.coreaudiotechnology.com/which-audiophile-playback-software-to-choose/ http://www.head-fi.org/t/617282/audirvana-plus-vs-bitperfect-vs-fidelia " 1. Sources of non-quality Assuming the output is bit-perfect, the computer as a source creates two main sources on non-quality: . Software-induced jitter Digital signal is in fact an analogue waveform composed of two states separated by a voltage threshold (1 if above, 0 if under). As presented in [MeitnerGendron91], the receiver detects the value change the moment the analogue value crosses the threshold. In addition, the shift from one state to another is not instantaneous but more slope like. So a slight change in the reference voltage of the source will lead to a slight temporal shift in the value change detection. So fluctuations in the source reference voltage create jitter, as explained in details in [HawksfordDunn96]. This is the same on the receiver side with measurement threshold fluctuations from its power supply and/or ground instability. Moreover the computer can still cause this as the grounds are linked most of the time through the same signal cables. Computer load means rapidly changing power demands from the CPU and its peripherals, with peak demands that are directly related to the software behaviour. . Radio-Frequency & other interferences In addition, computation, disk access, … activities mean complex current waveforms are carried on electrical lines and thus generate electromagnetic interferences. Apple computers are now made of “unibody” aluminium cases that are good protectors from inside RF interferences. But this is not sufficient as the cables connected to the computer act as antennas. And these current waveforms are also going back through the computer PSU, polluting the mains power supply. 4. The player software impact First of all the player should ensure bit-perfect reproduction of the signal by: . Adapting the DAC sample rate to each track native to avoid any unwanted sample rate conversion . Taking exclusive access ("hog mode") of the device to prevent other opened applications from interfering Furthermore, as we have seen in section 1, the computer load (and its variations) has an impact on sound quality. Minimizing such current demands and sources of interferences is key: . Loading tracks before playback (“memory play”) to reduce disk access and its audible, power and RFI impacts . Minimizing synchronous CPU load taken for the audio data streaming operations. In addition to reduce jitter, this also helps to reduce audible RF interferences patterns, especially in low frequencies." http://www.amr-audio.co.uk/html/dp777_tech-papers_OSX-Integermode.html http://www.head-fi.org/t/595071/android-phones-and-usb-dacs/6195#post_11264146 " Clock Jitter: how much is it important? Initially I started with the units positioned in the easiest way inside the box. Jitter measured at this point was 358ps (340-370ps). However, the clock case was near the power control chips of the transport, which appeared less than optimal. So I raised the clock just a few centimeters (5-6cm), leaving all the rest as above. Jitter dropped to 334ps (329-341ps). Moving the clock away from the board as much as possible given the short connection made the jitter utterly go down to 328ps (322-332ps). Having the clock output twisted pair go along the transport power chips had scarcely any effect. Then I moved the PSU back towards the main power supply output stages and cabling, but there was an immediate increase of jitter up to 346ps (338-351ps), so I moved the PSU into another position nearby: this quieter position was just above the main PSU rectifying diodes.... and don't ask me why this appeared a quieter position!" http://www.tnt-audio.com/clinica/jitter3_e.html http://www.head-fi.org/t/595071/android-phones-and-usb-dacs/6210#post_11270037 http://www.cse.psu.edu/~gcao/teach/458-02/c2.pdf "The myth that "0s are 0s and 1s are 1s -- it's all digital don't care about" is simply said wrong and misleading. It's not about just digital 1s or 0s actually. The data receiver must be able to read a certain analog voltage and need to declare it a 1 or 0. Since timing and shape of that bit and its distortion are continuously changing and usually far far away from being ideal, the receiver will see all but a clean rectangular evenly separated noise-free signal." http://soundcheck-audio.blogspot.co.uk/2011/11/touch-toolbox-30.html MikeyFresh and gixxerwimp like this.