Papers by Nikolaus Färber
Multimedia Communications, 1999
The heterogeneous structure of the Internet is a great obstacle for establishing real-time video ... more The heterogeneous structure of the Internet is a great obstacle for establishing real-time video services. Scalable video codecs, generating bit-streams decodable at di erent rates, have been proposed to address the heterogeneity problem. In this paper, we will review standardcompliant and non-compliant codec architectures for Internet video-ondemand. For compression based on the H.263 standard, we h a ve developed a compatible architecture that allows to switch b e t ween preencoded bit-streams of di erent bit-rates. This architecture provides excellent streaming performance for point-to-point communication scenarios that o er a low delay feedback c hannel. Then we present a non-compliant fully scalable video codec based on a spatio-temporal resolution pyramid that is available as a Netscape plugin for the World Wide Web. This approach can encode embedded lower bit-rate layers at the same overall bit-rate as needed by H.263 single-layer coding and can also support multicasting. Finally, w e show h o w scalable video coding can be e ciently combined with a retransmission protocol achieving graceful degradation and investigate the streaming performance of the above architectures.
Codage d'une pluralité de signaux d'informations à l'aide d'une puissance de calcul commune
Systeme de codage d'une pluralite de signaux d'informations a l'aide d'une puissa... more Systeme de codage d'une pluralite de signaux d'informations a l'aide d'une puissance de calcul commune, comprenant une pluralite de dispositifs de codage respectivement d'un signal des differents signaux d'informations a l'aide de la puissance de calcul commune, chaque dispositif de codage pouvant etre commande par au moins un parametre de codage respectif en ce qui concerne sa complexite de codage/son comportement en cas de perturbations de codage. Le systeme de codage comprend egalement un dispositif pour delivrer des informations, fonction des signaux, a chaque dispositif de codage, ces informations dependant du signal d'information respectif et indiquant une perturbation de codage du dispositif de codage respectif. Le systeme de codage comprend enfin un dispositif de reglage des parametres de codage en fonction des informations fonction des signaux en tenant compte de la puissance de calcul commune de telle facon qu'une combinaison de complexi...
Quality Scalable video signal, process for its production, the encoder and decoder
A quality-scalable video signal having an encoded using temporal prediction base signal having a ... more A quality-scalable video signal having an encoded using temporal prediction base signal having a base quality, from the from the quality-scalable video signal having a basic quality alone is reconstructed out and which has a first temporal repeat distance of intra-coded pictures, and an encoded using temporal prediction enhancement signal from which in combination with the base signal, the quality-scalable signal with improved quality can be reconstructed and having a second temporal repeat distance of intra-coded pictures, wherein the first temporal repeat distance is less than the second temporal Widerholabstand.
Enhancements in DVB-H and DVB-SH Based Mobile-TV Multiplexing
Microelectronic Systems, 2011
Abstract Energy efficient broadcasting of mobile-TV content through DVB-H and DVB-SH suffers from... more Abstract Energy efficient broadcasting of mobile-TV content through DVB-H and DVB-SH suffers from signaling limitations introduced by the DVB-H and DVB-SH multiplexing scheme resulting in an increased system delay. This paper presents a novel scheduling and ...
D2.2: Interim Reference Architecture Specification And Integration Report
Deliverable D2.2 provides an overview about the current state of the pilot implementation archite... more Deliverable D2.2 provides an overview about the current state of the pilot implementation architecture and its influence on the final reference architecture. Moreover, the integration activities so far are summarised. The reference architecture will be developed during the project time, based on experiences from the project pilots. A detailed explanation regarding the distinction between reference architecture and pilots is included. The planned workflow of the pilots is described as well as the current state of macroblocks and their components. This version of D2.2 includes updates, which were requested by the EC reviewer after the first submission. Hence, interfaces were identified and where necessary, described in more detail. This document will be updated once more during the project.
This deliverable describes the progress on representation, archiving and provision of object-base... more This deliverable describes the progress on representation, archiving and provision of object-based audio. It builds on D4.1 "Requirements for representation, archiving and provision of object-based audio". It lists the formats which are selected for ORPHEUS and describes the interim status of the implementation of these formats. On the production side, formats like BW64, ADM, NMOS, UMCP are used and explained. BW64 is also used for archiving. For provision or distribution MPEG-H and AAC + ADM metadata are selected. Both solutions use MPEG-DASH for streaming.<br> This Deliverables also serves as documentation for milestone MS12 "Initial implementation and documentation of a format for provision of objected-based audio" which has been achieved on 31/03/17.
D2.4: Final Reference Architecture Specification And Integration Report
This document describes the final reference architecture of ORPHEUS, a completely object-based, e... more This document describes the final reference architecture of ORPHEUS, a completely object-based, end-to-end broadcast and production workflow. It has been the subject of intensive discussions and several iterations over the duration of the project and has been shaped by considering typical channel-based broadcast workflows as well as the knowledge gained and lessons learned from the pilot phases. The architecture is format and interface agnostic and, as far as is possible, it should be applicable to a range of different infrastructures and ecosystems. Additionally, the pilot implementation and integration is also summarised.
2001 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings (Cat. No.01CH37221)
A new receiver-based playout scheduling scheme is proposed, which estimates the network delay fro... more A new receiver-based playout scheduling scheme is proposed, which estimates the network delay from past statistics and adaptively adjusts the playout time of the voice packets. In contrast to previous work, the adjustment is not only performed in between talkspurts, but also within the talkspurts in a highly dynamic way. Proper reconstruction of continuous output speech is achieved by scaling individual voice packets using a time-scale modification technique which modifies the rate of playout while preserving voice pitch. Subjective listening tests show that this operation does not impair audio quality. Simulation results based on Internet measurement indicate that buffering delay and loss rate can be significantly reduced by adaptive scheduling.
Feedback based error control for robust video transmission
1996 8th European Signal Processing Conference (EUSIPCO 1996), 1996
Video communication at very low bit-rates has made significant progress recently through the new ... more Video communication at very low bit-rates has made significant progress recently through the new ITU-T standard H.263. In this paper, we are reviewing the performance advances over the 1990 ITU-T standard H.261, and present a novel extension that allows robust transmission of moving video over highly unreliable channels, such as the mobile channel.
Advanced time shrinking using a drop classifier based on codec features
Interspeech 2015, 2015
We present an integrated approach of full-band audio time scale modification for Voice over IP co... more We present an integrated approach of full-band audio time scale modification for Voice over IP communication. The concept is based on a low complexity adaptive playout method that uses frame dropping and audio concealment for time shrinking and stretching, respectively. The existing version of this method is improved using a classifier that assists in choosing which audio frames can be dropped with the least subjective impact on audio quality. To maintain low complexity, we exclusively use audio signal features that are available in the audio codec. The classification of audio frames improves audio quality of the existing method without classification by 0.5 Mean Opinion Score points while requiring significantly less computational complexity by a factor of ca 10.
Interactivity-aware playout adaptation
Interspeech 2015, 2015
Adaptive Playout is a solution in IP-based communication clients to compensate network issues usi... more Adaptive Playout is a solution in IP-based communication clients to compensate network issues using a dynamic receiver buffer. Whereas small buffers provoke loss artefacts due to delayed packets, large buffers result in high delay and loss of interactivity. Existing Voice over IP clients balance the trade-off between low delay and loss artefacts based on empirical values. Most often the degree of interactivity is not taken into account and adaptation parameters remain the same for long monologues and lively discussions. We present a novel playout adaptation scheme that not only adapts to changing network conditions but also to conversational interactivity. Using an integrated approach we improve conversations with high interactivity significantly while at the same time preserve high audio quality. Using a full client implementation we verify our findings by conducting conversation tests and obtain gains of up to 0.9 Mean Opinion Score.
Audio Routing for Scalable Conferencing using AAC-ELD and Bit Stream Domain Energy Estimation
Proceedings of the 23rd ACM international conference on Multimedia - MM '15, 2015
There is an increasing interest in multipoint conferencing but service providers face the challen... more There is an increasing interest in multipoint conferencing but service providers face the challenge of complexity when scaling to thousands of users. This problem can be resolved by a scalable architecture based on central media routers, which allows for low complexity server components and therefore low operational cost. In this paper we describe an audio routing approach using advantages offered by the AAC-ELD bit stream structure. By estimating the signal energy in the bit stream domain, we can detect active speakers at low complexity, which results in substantial bitrate and complexity reduction. Subjective tests with mediated conversations among four participants are conducted to compare the perceived audio quality of the audio router to conventional mixing in a conference bridge. The results show a reduction of complexity by one order of magnitude while maintaining the same subjective audio quality when forwarding the two most active speakers on a 10 ms framing.
Error-Resilient Standard-Compliant Video Coding
... Laboratory University of Erlangen-Nuremberg Cauerstrasse 7, 91058 Erlangen, Germany Phone:+ 4... more ... Laboratory University of Erlangen-Nuremberg Cauerstrasse 7, 91058 Erlangen, Germany Phone:+ 49-9131-857101, Fax:+ 49-9131-858849 {girod, faerber}@ nt. e-technik. ... In practice, error tracking information will be stored in a cyclic data structure covering the last M frames. ...
<title>Adaptive optimal intra-update for lossy video transmission</title>
Visual Communications and Image Processing 2000, 2000
An adaptive algorithm to adjust the Intra update rate of a video encoder is presented. As optimiz... more An adaptive algorithm to adjust the Intra update rate of a video encoder is presented. As optimization criterion, the PSNR at the decoder after lossy transmission is used. Based on a model for the video encoder and error propagation at the video decoder the optimal Intra rate can be calculated for a given transmission channel analytically. In this paper it is shown how the parameters of these models can be measured adaptively during encoding. Thus, the Intra rate can be adjusted to changing channel conditions and sequence statistics. It is shown that a practical robust system can be built with the presented algorithm.
Internet Radio Surround: MPEG Surround over ISMA Ultravox
Optimizing channel change time in IPTV applications
2008 IEEE International Symposium on Broadband Multimedia Systems and Broadcasting, 2008
Abstract This paper describes the problems of channel change in multicast IPTV scenarios and the ... more Abstract This paper describes the problems of channel change in multicast IPTV scenarios and the sources of delay during channel change. It provides an overview of approaches to reduce the tune-in delay and discusses their impact on image quality and required bandwidth. ...
2011 Third International Conference on Communication Systems and Networks (COMSNETS 2011), 2011
We present an integrated approach for flexible playout adaptation for high quality audio transmis... more We present an integrated approach for flexible playout adaptation for high quality audio transmission over impaired network connections. The key concept of our framework is a continuous measurement of the transmission delay, the delay variation, and packet loss. Based on these measurements, the adaptive playout control employs audio time stretching using audio concealment and frame dropping techniques to keep the low delay requirements. In the literature, playout adaptation techniques have mainly been considered for voice over IP, using silence periods between talkspurts, or for high quality audio transmission over dedicated network links. To the best of our knowledge, our playout algorithm is the first achieving low delay high quality audio streaming over impaired network connections for both music and speech. We used a significant number of network traces to estimate the variation of the network quality in DSL, WLAN, UMTS and GPRS links and to update the parameters of our playout adaptation technique. Experimental results clearly indicate that our system provides very high accuracy for the desired accepted late loss rate and achieves a fast playout adaptation, even for rapidly changing network conditions.
Conversational quality as a function of delay and interactivity
The impact of mouth-to-ear delay on Conversational Quality (CQ) has been subject to manyfold rese... more The impact of mouth-to-ear delay on Conversational Quality (CQ) has been subject to manyfold research in the field of telecommunications. However, its dependency on Conversational Interactivity (CI) has not been studied explicitly in a quantitative way. To investigate the impact of both delay and CI, we conducted 48 tests with 96 different participants and controlled the interactivity through different tasks, namely natural conversation (flight), proof reading of text (text), and random number verification (rnv). The control of interactivity is proven to be effective and remains stable for delays from 100 ms to 800 ms. The Speaker Alternation Rate (SAR) is used as the metric for interactivity. It is shown to have a significant impact on CQ after 200ms delay and the effect does not alleviate until 800 ms. The hypothesis is proven using both graphical analysis and statistical evaluation based on a linear model and Analysis of Variance (ANOVA).
High-Definition Audio for Group-to-Group Communication
Microelectronic Systems, 2011
Abstract The European research project “Together anywhere, together anytime”(TA2) explores how te... more Abstract The European research project “Together anywhere, together anytime”(TA2) explores how technology can make communication and engagement easier among groups of people separated in space and time, eg families living in different cities. It combines ...
Uploads
Papers by Nikolaus Färber