Considering Bluetooth's Subband Codec (SBC) for Wideband Speech and Audio on the Internet

The Bluetooth Special Interest Group (SIG) has standardized the subband coding (SBC) audio codec to connect headphones via wireless Bluetooth links. SBC compresses audio at high fidelity while having an ultra-low algorithm delay. To make SBC suitable for the Internet, we extend it by using a time and packet loss concealment (PLC) algorithm that is based on ITU's G.711 Appendix I. The design is novel in the aspect of the interface between codec and speech receiver. We developed a new approach on how to distribute the functionality of a speech receiver between codec and application. Our approach leads to easier implementations of high quality VoIP applications. We conducted subjective and objective listening tests of the audio quality of SBC and PLC in order to determine an optimal coding mode and the trade-off between coding mode and packet loss rate. More precisely, we conducted MUSHRA listening tests for selected sample items. These tests results are then compared with the results of multiple objective assessment algorithms (ITU P.862 PESQ, ITU BS.1387-1 PEAQ, Creusere's algorithm). We found out that a combination of the PEAQ basic and advanced values best matches---after third order linear regression---the subjective MUSHRA results . The linear regression has coefficient of determination of R²=0.907². By comparison, our individual human ratings show a correlation of about R=0.9 compared to our averaged human rating results. Using the combination of both PEAQ algorithms, we calculate hundred thousands of objective audio quality ratings varying audio content and algorithmic parameters of SBC and PLC. The results show which set of parameters value are best suitable for a bandwidth and delay constrained link. The transmission quality of SBC is enhanced significantly by selecting optimal encoding parameters as compared to the default parameter sets given in the standard. Finally, we present preliminary objective tests results on the comparison of the audio codecs SBC, CELT, APT-X and ULD coding speech and audio transmission. They all allow a mono and stereo transmission of music at ultra-low coding delays (<10ms), which is especially useful for distributed ensemble performances over the Internet

An Architecture for a Next Generation VoIP Transmission System

Dieser Beitrag ist mit Zustimmung des Rechteinhabers aufgrund einer (DFG geförderten) Allianz- bzw. Nationallizenz frei zugänglich.This publication is with permission of the rights owner freely accessible due to an Alliance licence and a national licence (funded by the DFG, German Research Foundation) respectively.Packetized speech transmission systems implemented with Voice over IP are gaining momentum against the traditional circuit switched systems despite the fact that packet switched VoIP is two to three times less efficient than its circuit switched counterpart. At the same time, it only supports a rather bad “toll” quality. We believe that it is time for a new architecture developed from scratch – an architecture that includes an Internet enabled speech codec and its transport system. This architecture manages the perceptual service quality while using the available transmission resources to its best. The transmission of speech is managed and controlled with respect to its speech quality, mouth-to-ear delay, bit-rate, frame-rate, and loss robustness. Beside the architecture, we describe the requirements for the Internet speech codec and its transport protocol and present an interface between the speech codec and the transport protocol

Technical and Clinical Outcome of Talent versus Endurant Endografts for Endovascular Aortic Aneurysm Repair

The technical evolution of endografts for the interventional management of infrarenal abdominal aortic aneurysms (AAA) has allowed a continuous expansion of indications. This study compares the established Talent endograft with its successor, the Endurant endograft, taking individual aortoiliac anatomy into account.From June 2007 to December 2010, 35 patients with AAA were treated with a Talent endograft (33 men) and 36 patients with an Endurant endograft (34 men). Aortoiliac anatomy was evaluated in detail using preinterventional computed tomography angiography. The 30-day outcome of both groups were compared regarding technical and clinical success as well as complications including endoleaks.The Endurant group included more patients with unfavorable anatomy (kinking of pelvic arteries, p = 0.017; shorter proximal neck, p = 0.084). Primary technical success was 91.4% in the Talent group and 100% in the Endurant group (p = 0.115). Type 1 endoleaks occurred in 5.7% of patients in the Talent group and in 2.8% of those in the Endurant group (p = 0.614). Type 3 endoleaks only occurred in the Talent group (2.9% of patients; p = 0.493). Type 2 endoleaks were significantly less common in the Endurant group than in the Talent group (8.3% versus 28.6%; p = 0.035). Rates of major and minor complications were not significantly different between both groups. Primary clinical success was significantly better in the Endurant group (97.2%) than in the Talent group (80.0%) (p = 0.028).Endurant endografts appear to have better technical and clinical outcome in patients with difficult aortoiliac anatomy, significantly reducing the occurrence of type 2 endoleaks

Internet Telephonie über drahtlose Verbindungen

Viele Telefongespräche werden über IP-basierte Kommunikationssyste wie WLAN, Hyperlan oder Wimax vermittelt. Diese Arbeit beschreibt Algorithmen, die die Effizienz der Sprachübertragung erhöhen. Wir verfolgen dabei das Ziel, die vom Menschen wahrgenommene Qualität zu optimieren, aber zugleich Normierungen bestehender und zukünftiger Kommunikationssysteme einzuhalten. Wir stellen einen verifizierten Algorithmus vor, der die wahrgenomme Qualität von Internet Telefongesprächen bewertet. Mit unserem Ansatz leiten wir darüber hinaus wichtige Designentscheidungen für Transport- und Vermittlungsschichtprotokolle analytisch her. Wenn hochgradig komprimierte Sprachdaten über das Internet übertragen werden, kann der Verlust eines Paketes sehr unterschiedliche Auswirkungen haben. Dies hängt unter anderem von dem Inhalt und Kontext des Sprachpakets ab. Wir definieren die Paketwichtigkeit als die Verschlechterung der Sprachqualität nach Verlust eines Sprachpakets. Wir entwickelten Verfahren, welche die Paketwichtigkeiten offline und in Echtzeit quantitativ bestimmen. Wendet man Paketwichtigkeiten an, zeigt sich, dass nur ein Bruchteil aller Sprachpakete übertragen werden muss, wenn zumindest Sprachverständlichkeit gewährleistet werden soll. Wird das Konzept der Paketwichtigkeiten bei drahtlosen Internet-Telefonsystemen angewendet, läßt sich der Energieverbrauch der Funktelefone signifikant reduzieren, da weniger Pakete übertragen werden müssen. Schlussendlich zeigen wir, dass die Entfernung zwischen zwei Wi-Fi Geräten einfach und genau gemessen werden kann, indem man die Laufzeiten der Pakete bestimmt. Unser Verfahren stellte sich den bisher verwendeten Feldstärkemessungen als überlegen heraus. Zusammenfassend kann man sagen, dass diese Arbeit neue Algorithmen und relevante Innovationen enthält, deren Einfluss auf Forschungen und Produktentwicklung sich in den nächsten Jahren erst voll zeigen wird.Many telephony calls are carried over IP-based technologies, which include wireless systems such as WLAN, Hyperlan or Wimax. This thesis contains algorithms which reduce the overhead of IP-based speech communication. The aim is to increase the human perceived call quality while recognising communication standards defining Internet telephony and wireless systems. We present and verify an instrumental approach on how to assess the perceptual quality of voice transmissions in IP-based communication networks. Using our approach we derivate by analysis design guidelines for application and data-link protocols. If highly compressed, packetized speech is transported over packet networks, losses of individual packets impair the perceptual quality of the received stream by different degrees, depending on the content and context of the lost packets. We introduce the idea of the Importance of Individual Packets, which is defined by impact of packet loss on speech quality. We present real-time and off-line algorithms to measure the importance. Using the concept of importance of packets we show that only a fraction of all speech packets need to be transmitted if speech intelligibility is to be maintained. If this concept is applied for Internet telephony over wireless links, significant transmission energy savings on wireless phones can be achieved, because fewer packets need to be transmitted. Last, we proved that the distance between two WLAN nodes can be determined by packet round trip time measurements. This approach outperforms the previously used signal strength indications. Overall, this thesis contains relevant innovations and proposes novel algorithms that will have a substantial impact on research and future product development

Abstract Classifying VoIP µ-law Packets in Real-Time

We present an algorithm, which classifies µ-law coded VoIP packets in real time. It is based on our algorithm calculating frame importance off-line, but uses shorter speech segments, an analysis-by-synthesis approach, and a newly developed perceptual evaluation algorithm called PESQlight. Altogether, we are able to reduce the complexity and algorithm delay significantly. The real-time classification has a correlation of up to R=0.63 compared to the reference, off-line algorithm. 1

Error Propagation After Concealing a Lost Speech Frame

Abstract — Depending on the content of speech frames, the quality impairment after their loss differs widely. In previous publications we described an off-line measurement procedure to determine the loss impairment – the importance – of single speech frames. We showed that knowing the importance of frames can enhance the transmission performance of VoIP telephones significantly if only important frames are transmitted. Here we study to what extend the importance can be calculated at real-time: The loss impairment is due to the imperfect packet loss concealment (PLC) and also due to error propagation (EP). EP originates from the desynchronisation of the decoder’s internal state and cannot be calculated at real-time. We developed a measurement method to determine the effect of the imperfect PLC and the temporal progression of the error propagation. The results show the trade-off between algorithmic delay and the accuracy of real-time importance calculation: A good frame classification needs to look ahead 20-40 ms in order to calculate the importance precisely. I