# Impact of Audio Delay and Quality in Network Music Performance +
## 3 Experimental Setup  
For our experiments, we utilized two acoustically and visually isolated rooms located on the same floor of the main building at the Athens University of Economics and Business. Musicians performed in separate rooms, communicating via headphones for audio and 32\" monitors for video. We investigated two experimental conditions: in Scenario A, we varied the audio delay while maintaining constant audio quality, whereas in Scenario B, we varied the audio quality while keeping the audio delay fixed. The overall setup remained similar across both scenarios, with slight modifications depending on the parameters being tested. In Scenario A, illustrated in Figure 1, each room was equipped with an 8-channel audio mixer responsible for routing, monitoring, and recording audio. Audio was captured using condenser microphones, and monitoring was achieved through closed-type headphones. Video was recorded using a camera in each room and transmitted as a composite analog video signal to a 32\({}^{\prime\prime}\) display in the other room (represented by red lines in the figure). The network cables from both rooms were connected directly to each other, using a twisted pair from a UTP bundle to carry the analog composite video signal, bypassing any switching equipment. The use of composite video aimed to minimize visual latency. Analog video transmission displays content line-by-line, avoiding the buffering and processing delays inherent in digital video, which must capture, encode, transmit, decode, and render full frames. To measure round-trip video latency, we positioned a smartphone displaying a running timer in front of the video camera in one room and directed the camera in the other room at its monitor, thereby reflecting the timer image back. A separate camera recorded both the original and reflected timers. Frame-by-frame analysis of the recording showed a round-trip video delay of 30 ms, corresponding to a one-way delay of approximately 15 ms. The audio mixers in both rooms were also connected via direct analog cabling, avoiding digital processing altogether. This configuration enabled us to achieve precise, low-latency audio transmission, with fixed delays even below 10 ms, something not feasible when routing through digital devices or switches. To control delay, we used two AD-340 analog audio delay units by Audio Research, allowing us to set exact delay values in each direction. In Scenario B, the configuration was modified as shown in Figure 2. Audio outputs from the mixers were routed into Linux-based PCs (Intel i7 CPUs, 12 GB RAM), which ran our custom NMP software, Aretousa; this software is available upon request from the authors. The software was used solely to adjust the audio sampling rate. Importantly, no compression or additional signal processing was applied. The video setup remained identical to Scenario A, preserving the one-way delay of 15 ms. PCs were interconnected via the building's Fast Ethernet LAN, which included three Ethernet switches along the path. Under these conditions, we measured a one-way audio delay of approximately 10 ms. Most studies on the effects of delay on NMP focus on the one-way delay between two participants; in an audio context, we call this the Mouth to Ear (M2E) delay, as it represents the delay from the microphone of one musician to the headphones of the other musician. In practice, M2E delay is extremely hard to measure, as it requires very fine clock synchronization between the endpoints, which is only feasible with sophisticated equipment [25]. For this reason, in this work we measured My Mouth to My Ear (MM2 ME) delay, which is basically the two-way version of M2E, as shown in Figure 3. MM2 ME delay is easy to measure accurately, by simply reflecting the audio signal and then looking at  
Figure 1: Scenario A experimental setup (variable delay). the outgoing and incoming signal in an audio analysis application.