J.C. De Martin

Perception-based partial encryption of compressed speech

Mobile multimedia applications, the focus of many forthcoming wireless services, increasingly demand low-power techniques implementing content protection and customer privacy. In this paper low complexity perception-based partial encryption schemes for speech are presented. Speech compressed by a widely-used speech coding algorithm, the ITU-T G.729 standard at 8 kb/s, is partitioned in two classes, one, the most perceptually relevant, to be encrypted, the other, to be left unprotected. Two partial-encryption techniques are developed, a low-protection scheme, aimed at preventing most kinds of eavesdropping and a high-protection scheme, based on the encryption of a larger share of perceptually important bits and meant to perform as well as full encryption of the compressed bitstream. The high-protection scheme, based on the encryption of about 45% of the bitstream, achieves content protection comparable to that obtained by full encryption, as verified by both objective measures and formal listening tests. For the low-protection scheme, encryption of as little as 30% of the bitstream virtually eliminates intelligibility as well as most of the remaining perceptual information. Low-power, portable devices could therefore achieve very high levels of speech-content protection at only 30-45% of the computational load of current techniques, freeing resources for other tasks and enabling longer battery life.

A simulation study of adaptive voice communications on IP networks

This paper presents simulation results outlining the behavior of rate-adaptive voice communications over IP networks. In the considered architecture, voice coders adapt their rate to the current state of the network so as to generate only the bandwidth that the network is capable of carrying. An algorithm is proposed for driving the transmission rate of voice sources on the basis of estimations of the network conditions, measured in terms of packet delays and losses. The effectiveness of the proposed solution is then investigated in various scenarios which comprise: (i) a dedicated network in which the available bandwidth is exclusively shared between adaptive voice connections; (ii) a scenario in which adaptive voice sources compete with other TCP-like sources; and (iii) an uncontrolled network environment. We have compared the performance of the rate-adaptive against the non-adaptive (i.e. fixed-rate) approach for the transport of voice over IP. Using a rate-adaptive approach, more voice communications can be carried while maintaining a good quality of service, even on non-segregated networks.

Analysis-by-synthesis distortion computation for rate-distortion optimized multimedia streaming

This paper presents an analysis-by-synthesis technique to evaluate the perceptual importance of multimedia packets for rate-distortion optimized streaming. The proposed technique, instead of relying on a priori information, computes the distortion that would be caused by the loss of each single packet, including the effects of error propagation and receiver-side error concealment. A rate-distortion optimized streaming algorithm is presented to compare the perceptual performance obtained using content-adaptive analysis-by-synthesis distortion values versus distortion values obtained using a priori knowledge of the statistical importance of the elements of the compressed multimedia bitstream. Simulations with video test sequences compressed with the MPEG-2 coding standard show that the proposed technique delivers substantial and consistent PSNR gains (1.2-2.8 dB) with respect to ideal frame type-driven a priori distortion evaluation for a wide range of channel conditions. Compared to distortion-agnostic streaming techniques such as SoftARQ, the gain is even more pronounced.

Selective encryption and scalable speech coding for voice communications over multi-hop wireless links

With the rapid deployment of wireless LAN and the future development of mobile ad hoc networks, multi-hop wireless communications links are expected to become much more common. How to achieve reliable and efficient, yet secure, voice communication over such multi-hop wireless links is an important issue. This paper proposes and investigates a combination of scalable speech coding and selective encryption for secure voice communication over multi-hop wireless links that addresses both the efficient use of network and node resources and security against unwanted eavesdroppers. It is shown that when the Shannon lower bound is satisfied with equality for rate distortion optimal scalable coding, transmission of the enhancement layer in-the-clear provides no information regarding the core layer. A specific example of combining selective encryption with MPEG-4 scalable speech coding demonstrates the effectiveness and efficiency of the approach.

Real-Time Transmission of H.264 Video Over 802.11b-Based Wireless Ad Hoc Networks

This chapter aims at evaluating a number of Quality of Service (QoS) indices of a real-time video transmission over an 802.11b ad hoc wireless network. Video is coded according to the state-of-the-art ITU-T H.264 encoder and its transmission is simulated by means of the ns-2 network simulator. Objective quality measurements are presented. Moreover, the impact of different parameters — both at the encoder and at the MAC level —, of background interfering traffic and of the number of relay nodes, is studied, showing the various trade-offs involved.

Distortion-based packet marking for mpeg video transmission over diffserv networks

We present a distortion-based approach to packet classification for multimedia transmission over differentiatedservices packet networks. Instead of sending all traffic as premium or relying on a priori data partitioning, packets are individually examined and assigned to different service classes depending on the level of distortion that their loss would introduce at the decoder. Applied to video sequences encoded with the ISO MPEG-2 video coding standard, the proposed distortion-based packet marking scheme outperforms source-transparent techniques and provides substantial and consistent gains in PSNR over the regular best-effort case sending as little as 10% of the packets as premium traffic. Video samples are available at

Cross-layer perceptual ARQ for video communications over 802.11e wireless networks

This work presents an application-level perceptual ARQ algorithm for video streaming over 802.11e wireless networks. A simple and effective formula is proposed to combine the perceptual and temporal importance of each packet into a single priority value, which is then used to drive the packet-selection process at each retransmission opportunity. Compared to the standard 802.11 MAC-layer ARQ scheme, the proposed technique delivers higher perceptual quality because it can retransmit only the most perceptually important packets reducing retransmission bandwidth waste. Video streaming of H.264 test sequences has been simulated with ns in a realistic 802.11e home scenario, in which the various kinds of traffic flows have been assigned to different 802.11e access categories according to the Wi-Fi alliance WMM specification. Extensive simulations show that the proposed method consistently outperforms the standard link-layer 802.11 retransmission scheme, delivering PSNR gains up to 12 dB while achieving low transmission delay and limited impact on concurrent traffic. Moreover, comparisons with a MAC-level ARQ scheme which adapts the retry limit to the type of frame contained in packets and with an application-level deadline-based priority retransmission scheme show that the PSNR gain offered by the proposed algorithm is significant, up to 5 dB. Additional results obtained in a scenario in which the transmission relies on an intermediate node (i.e., the access point) further confirms the consistency of the perceptual ARQ performance. Finally, results obtained by varying network conditions such as congestion and channel noise levels show the consistency of the improvements achieved by the proposed algorithm.

MAC-level partial checksum for H.264 video transmission over 802.11 ad hoc wireless networks

We present a technique to enhance real-time video transmissions over 802.11-based ad hoc networks. We propose to limit the coverage of the error detection mechanism to the most error sensitive bit class contained in video packets. Packets are retransmitted only if errors affect those bits, instead of retransmitting all corrupted packets irrespective of error positions. The expectation was that the negative effects of errors in the less sensitive bits should be more than adequately counterbalanced by the lower number of discarded video packets. Moreover, the lower number of retransmissions should reduce network load, with positive effects on all the transmissions. We simulated the transmission of H.264 video over an ad hoc scenario with a varying number of relay nodes using ns with experimental bit error traces. Results show consistent video quality improvements (up to 1 dB PSNR) as well as network load reduction with respect to state-of-the-art techniques, such as unequal error protection based on different retry limits.

Frequency-selective partial encryption of compressed audio

The widespread adoption of compressed digital audio increasingly demands effective ways to conjugate ease of distribution with content protection. In this paper a low-complexity partial-encryption scheme for MPEG audio is presented. The aim is to provide listeners with sample-quality audio material that can be upgraded to full-quality by simply acquiring a key and decrypting a few selected bits (1-10% of the total bitstream) of the already available data. Sample-quality is obtained by limiting the frequency content of the signal, and it is achieved without altering the compatibility of the compressed audio bitstream. Audio material partially encrypted with the proposed scheme can thus be freely distributed for evaluation and then easily unlocked to achieve full quality without any further transmission of audio data.

Delivery of MPEG video streams with constant perceptual quality of service

Constant levels of perceptual quality of service is what ideally users of multimedia services expect. In most cases, however, they receive time-varying levels of quality of service. This paper describes a technique to deliver nearly constant perceptual quality of service when transmitting video sequences over differentiated services IP networks. MPEG video packets are transmitted either as low-loss premium packets or as regular best-effort packets depending on their individual perceptual importance. On a frame-by-frame basis, allocation to the premium class is performed depending on the perceptual importance of each macroblock, the desired level of quality of service and the instantaneous network state. The resulting perceptually-based, time-varying use of premium and best-effort network resources delivers nearly constant quality of service to end users and it yields significant higher PSNR values compared to constant allocation of premium bandwidth.