Using Autoregressive Models for Real-Time Packet Loss Concealment in Networked Music Performance Applications

In Networked Music Performances (NMP), concealing the effects of lost/late packets on the quality of the playback audio stream is of pivotal importance to mitigate the impact of the resulting audio artifacts. Traditional packet loss concealment techniques implemented in standard audio codecs can be leveraged only at the price of an increased mouth-to-ear latency, which may easily exceed the strict delay requirements of NMP interactions. This paper investigates the adoption of a low-complexity prediction technique based on autoregressive models to fill audio gaps caused by missing packets. Numerical results show that the proposed approach outperforms packet loss concealment methods normally implemented in NMP systems, typically based on filling audio gaps with silence or repetition of the last received audio segment

Covert Communication Networks

A covert communications network (CCN) is a connected, overlay peer-to-peer network used to support communications within a group in which the survival of the group depends on the confidentiality and anonymity of communications, on concealment of participation in the network to both other members of the group and external eavesdroppers, and finally on resilience against disconnection. In this dissertation, we describe the challenges and requirements for such a system. We consider the topologies of resilient covert communications networks that: (1) minimize the impact on the network in the event of a subverted node; and (2) maximize the connectivity of the survivor network with the removal of the subverted node and its closed neighborhood. We analyze the properties of resilient covert networks, propose measurements for determining the suitability of a topology for use in a covert communication network, and determine the properties of an optimal covert network topology. We analyze multiple topologies and identify two constructions that are capable of generating optimal topologies. We then extend these constructions to produce near-optimal topologies that can “grow” as new nodes join the network. We also address protocols for membership management and routing. Finally, we describe the architecture of a prototype system for instantiating a CCN

On-line monitoring of VoIP quality using IPFIX

The main goal of VoIP services is to provide a reliable and high-quality voice transmission over packet networks. In order to prove the quality of VoIP transmission, several approaches were designed. In our approach, we are concerned about on-line monitoring of RTP and RTCP traffic. Based on these data, we are able to compute main VoIP quality metrics including jitter, delay, packet loss, and finally R-factor and MOS values. This technique of VoIP quality measuring can be directly incorporated into IPFIX monitoring framework where an IPFIX probe analyses RTP/RTCP packets, computes VoIP quality metrics, and adds these metrics into extended IPFIX flow records. Then, these extended data are stored in a central IPFIX monitoring system called collector where can be used for monitoring purposes. This paper presents a functional implementation of IPFIX plugin for VoIP quality measurement and compares the results with results obtained by other tools

Quality aspects of Internet telephony

Internet telephony has had a tremendous impact on how people communicate. Many now maintain contact using some form of Internet telephony. Therefore the motivation for this work has been to address the quality aspects of real-world Internet telephony for both fixed and wireless telecommunication. The focus has been on the quality aspects of voice communication, since poor quality leads often to user dissatisfaction. The scope of the work has been broad in order to address the main factors within IP-based voice communication. The first four chapters of this dissertation constitute the background material. The first chapter outlines where Internet telephony is deployed today. It also motivates the topics and techniques used in this research. The second chapter provides the background on Internet telephony including signalling, speech coding and voice Internetworking. The third chapter focuses solely on quality measures for packetised voice systems and finally the fourth chapter is devoted to the history of voice research. The appendix of this dissertation constitutes the research contributions. It includes an examination of the access network, focusing on how calls are multiplexed in wired and wireless systems. Subsequently in the wireless case, we consider how to handover calls from 802.11 networks to the cellular infrastructure. We then consider the Internet backbone where most of our work is devoted to measurements specifically for Internet telephony. The applications of these measurements have been estimating telephony arrival processes, measuring call quality, and quantifying the trend in Internet telephony quality over several years. We also consider the end systems, since they are responsible for reconstructing a voice stream given loss and delay constraints. Finally we estimate voice quality using the ITU proposal PESQ and the packet loss process. The main contribution of this work is a systematic examination of Internet telephony. We describe several methods to enable adaptable solutions for maintaining consistent voice quality. We have also found that relatively small technical changes can lead to substantial user quality improvements. A second contribution of this work is a suite of software tools designed to ascertain voice quality in IP networks. Some of these tools are in use within commercial systems today

Synchronization of streamed audio between multiple playback devices over an unmanaged IP network

When designing and implementing a prototype supporting inter-destination media synchronization – synchronized playback between multiple devices receiving the same stream – there are a lot of aspects that need to be considered, especially when working with unmanaged networks. Not only is a proper streaming protocol essential, but also a way to obtain and maintain the synchronization of the clocks of the devices. The thesis had a few constraints, namely that the server producing the stream should be written for the .NET-platform and that the clients receiving it should be using the media framework GStreamer. This framework provides methods for both achieving synchronization as well as resynchronization. As the provided resynchro- nization methods introduced distortions in the audio, an alternative method was implemented. This method focused on minimizing the distortions, thus maintain- ing a smooth playback. After the prototype had been implemented, it was tested to see how well it performed under the influence of packet loss and delay. The accuracy of the synchronization was also tested under optimal conditions using two different time synchronization protocols. What could be concluded from this was that a good synchronization could be maintained on unloaded networks using the proposed method, but when introducing delay the prototype struggled more. This was mainly due to the usage of the Network Time Protocol (NTP), which is known to perform badly on networks with asymmetric paths.When working with synchronized playback it is not enough just obtain- ing it – it also needs to be maintained. Implementing a prototype thus involves many parts ranging from choosing a proper streaming protocol, to handling glitch free resynchronization of audio. Synchronization between multiple speakers has a wide area of application, ranging from home entertainment solutions to big malls where announcements should appear synchronized over the entire perimeter. In order to achieve this, two main parts are involved: the streaming of the audio, and the actual synchronization. The streaming itself poses problems mostly since the prototype should not only work on dedicated networks, but rather on all kinds, such as the Internet. As the information over these networks are transmitted in packets, and the path from source to destination crosses many sub networks, the packets may be delayed or even lost. This may create an audible distortion in the playback. The next part is the synchronization. This is most easily achieved by putting a time on each packet stating when in the future it should be played out. If then all receivers play it back at the specified time, synchronization is achieved. This however requires that all the receivers share the idea of when a specific time is – the clocks at all the receivers must be synchronized. By using existing software and hardware solutions, such as the Network Time Protocol (NTP) or the Precision Time Protocol (PTP), this can be accomplished. The accuracy of the synchronization is therefore partly dependent on how well these solutions work. Another valid aspect is how accurate the synchronization must be for the sound to be perceived as synchronized by humans. This is usually in the range of a few tens of milliseconds to five milliseconds depending on the sound. When a global time has been distributed to all receivers, matters get more complicated as there is more than one clock to consider at each receiver. Apart from the previously mentioned clock, now called the ’system clock’, there is also an audio clock, which is a hardware clock positioned on the sound card. This audio clock decides the rate at which media is played out. Altering the system clock to synchronize it to a common time is one thing, but altering the audio clock while media is being played will inevitably mean a jump in the playback, and thus a distortion. Although an initial synchronization can be achieved, the two clocks will over time tick in slightly different pace, thus drifting away from each other. This creates a need for the audio clock to continuously correct itself to follow the system clock. In the media framework GStreamer, used for handling the media at the re- ceivers, two alternatives to solve the correction problem were available. Quick evaluations of these two methods however showed that either audible glitches or ’oscillations’ occurred in the sound, when the clocks were corrected. A new method, which basically combines the two existing, was therefore implemented. With this method the audio clock is continuously corrected, but in a smaller and less aggressive way. Listening tests revealed much smaller, often not audible, distortions, while the synchronization performance was at par with the existing methods. More thorough testing showed that the synchronization over networks with light traffic was in the microsecond-range, thus far below the threshold of what will appear as synchronized. During worse conditions – simulated hostile environments – the synchronization quickly reached unacceptable levels though. This was due to the previously mentioned NTP, and not the implemented method on the other hand

Understanding How Agricultural Intensification Impacts Ring-Necked Pheasant Distribution and Survival in Eastern South Dakota

Historically, pheasants (Phasianus colchicus) have thrived in South Dakota in conjunction with successful land retirement programs or early farming practices through the 1950s, which created interspersions of agriculture and native landscapes that were ideal for pheasants. Recently, the Prairie Pothole ecosystem has undergone rapid agroeconomic expansion, effectively reducing ideal interspersions of native prairie and cropland into agriculturally homogeneous landscapes. Indices of pheasant abundance have suggested persistent population declines since 2008, raising concerns regarding landscape suitability. Our goal was to understand how agriculture intensification impacts pheasant ecology. The objectives were to: 1) estimate overwinter hen probability of survival, resource selection, and mortality risks associated with landscape features; 2) determine pheasant abundance as a function of remotely derived landscape composition and vegetative phenology; and 3) implement low-cost Arduino GPS trackers into our ring-necked pheasant study to improve fine-scale data collection. To accomplish these goals, we captured, collared, and tracked 100 female pheasants annually from December through March in each of three years (2016–2019). Overall, we captured 321 females and recorded 110 mortalities. We implemented low-cost GPS trackers on 35 individuals, resulting in 407 VHF locations and 1,574 GPS locations. This was a 287% increase in data density at only 23% increase in cost. We modeled weekly probability of survival and Cox proportional-hazard cause-specific mortality rates associated with landscape features. To understand pheasant distribution, we surveyed for and modeled pheasant abundance and distribution seasonally as a function of landscape composition and intraannual differences in vegetation phenology. Overwinter survival of pheasants (0.66) was highly influenced by snow depth. Pheasants using harvested fields experienced a 421% increase in risk of raptor predation. Additionally, pheasants using emergent wetlands experienced a 58% lower risk of weather mortality. Our distribution model demonstrated that proportion of Conservation Reserve Program grasslands, dormant wetlands, and 30- 40% row-crop agriculture within 1.6 km2 positively influenced pheasant abundance. Alternatively, pheasants were negatively associated with proportion of forests. Agricultural intensification is projected to continue reducing valuable concealment, grassland, and emergent wetland landscapes. As native perennial vegetation is critical to both pheasant abundance and winter survival, large-scale conservation efforts are critical to pheasant population viability. Innovative conservation solutions supplementing current farm bill policies may improve conservation adoption thereby improving pheasant abundance and overwinter survival

Review of Research on Speech Technology: Main Contributions From Spanish Research Groups

In the last two decades, there has been an important increase in research on speech technology in Spain, mainly due to a higher level of funding from European, Spanish and local institutions and also due to a growing interest in these technologies for developing new services and applications. This paper provides a review of the main areas of speech technology addressed by research groups in Spain, their main contributions in the recent years and the main focus of interest these days. This description is classified in five main areas: audio processing including speech, speaker characterization, speech and language processing, text to speech conversion and spoken language applications. This paper also introduces the Spanish Network of Speech Technologies (RTTH. Red Temática en Tecnologías del Habla) as the research network that includes almost all the researchers working in this area, presenting some figures, its objectives and its main activities developed in the last years