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An interactive audio service is a new conceptual audio service that provides the users with opportunities for a variety of experiences on the alternative and advanced audio services. In the interactive audio service, users can freely control various audio objects to make their own audio sounds. A spatial audio object coding (SAOC) is a useful technology that can support most parts of the interactive audio service with a relatively low bit-rate, but is very poor to perfect gain control of a certain audio object, i.e., the target audio object. In this paper, the SAOC with a two-step coding structure is proposed to efficiently handle the target audio object as well as the normal audio objects. A transform coded excitation (TCX) based residual coding scheme is presented in the context of the sound quality enhancement. From experimental results, it can be noted that the various audio objects can be successfully handled with respect to the bit-rate and the sound quality by using the proposed two-step coding structure SAOC. [ABSTRACT FROM PUBLISHER]

Audio scrambling can be employed to ensure confidentiality in audio distribution. We first describe scrambling for raw audio using the discrete wavelet transform (DWT) first and then focus on MP3 audio scrambling. We perform scrambling based on a set of keys which allows for a set of audio outputs having different qualities. During descrambling, the number of keys provided and the number of rounds of descrambling performed will decide the audio output quality. We also perform scrambling by using multiple keys on the MP3 audio format. With a subset of keys, we can descramble to obtain a low quality audio. However, we can obtain the original quality audio by using all of the keys. Our experiments show that the proposed algorithms are effective, fast, simple to implement while providing flexible control over the progressive quality of the audio output. The security level provided by the scheme is sufficient for protecting MP3 music content. [ABSTRACT FROM PUBLISHER]

In audio watermarking area, the robustness against desynchronization attacks, such as TSM (Time-Scale Modification) and random cropping operations, is still one of the most challenging issues. In this paper, we present a multibit robust audio watermarking solution for such a problem by using the insensitivity of the audio histogram shape and the modified mean to TSM and cropping operations. We address the insensitivity property in both mathematical analysis and experimental testing by representing the histogram shape as the relative relations in the number of samples among groups of three neighboring bins. By reassigning the number of samples in groups of three neighboring bins, the watermark sequence is successfully embedded. In the embedding process, the histogram is extracted from a selected amplitude range by referring to the mean in such a way that the watermark will be able to be resistant to amplitude scaling and avoid exhaustive search in the extraction process. The watermarked audio signal is perceptibly similar to the original one. Experimental results demonstrate that the hidden message is very robust to TSM and random cropping attacks, and also has a satisfactory robustness for those common audio signal processing operations. [ABSTRACT FROM PUBLISHER]

We consider streaming pre-encoded and packetized media over best-effort networks in the presence of acknowledgment feedbacks. We first review a rate-distortion (RD) optimization framework that can be employed in such scenarios. As part of the framework, a scheduling algorithm selects the data to send over the network at any given time, so as to minimize the end-to-end distortion, given an estimate of channel resources and a history of previous transmissions and received acknowledgements. In practice, a greedy scheduling strategy is often considered to limit the solution search space, and reduce the computational complexity associated to the RD optimization framework. Our work observes that popular greedy schedulers are strongly penalized by early retransmissions. Therefore, we propose a scheduling algorithm that avoids premature retransmissions, while preserving the low computational complexity aspect of the greedy paradigm. Such a scheduling strategy maintains close to optimal RD performance when adapting to network bandwidth fluctuations. Our experimental results demonstrate that the proposed patient greedy scheduler provides a reduction of up to 50% in transmission rate relative to conventional greedy approaches, and that it brings up to 2 dB of quality improvement in scheduling classical MPEG-based packet video streams [ABSTRACT FROM PUBLISHER]

This work proposes an audio data hiding system that hides information into signals not known beforehand. The system can hide data into live music, or ambient sounds in general, and can be used to communicate information acoustically from one device to another. An important benefit of such a system is backward-compatibility, as the transmitter is a speaker, and the receiver is a microphone, both of which are already present in numerous devices and environments. The highest data rate achieved was 213 bits/s, while keeping the error rate under 10%. The developed technique was applied in a simple navigation system, where acoustic data embedded into background music indicates the location of the receiver [ABSTRACT FROM PUBLISHER]

This paper studies the quality of multimedia content at very low bitrates. We carried out subjective experiments for assessing audiovisual, audio-only, and video-only quality. We selected content and encoding parameters that are typical of mobile applications. Our focus were the MPEG-4 AVC (a.k.a. H.264) and AAC coding standards. Based on these data, we first analyze the influence of video and audio coding parameters on quality. We investigate the optimal trade-off between bits allocated to audio and to video under global bitrate constraints. Finally, we explore models for the interactions between audio and video in terms of perceived audiovisual quality [ABSTRACT FROM PUBLISHER]

This paper addresses the problem of streaming packetized media over a lossy packet network in a rate-distortion optimized way. We show that although the data units in a media presentation generally depend on each other according to a directed acyclic graph, the problem of rate-distortion optimized streaming of an entire presentation can be reduced to the problem of error-cost optimized transmission of an isolated data unit. We show how to solve the latter problem in a variety of scenarios, including the important common scenario of sender-driven streaming with feedback over a best-effort network, which we couch in the framework of Markov decision processes. We derive a fast practical algorithm for nearly optimal streaming in this scenario, and we derive a general purpose iterative descent algorithm for locally optimal streaming in arbitrary scenarios. Experimental results show that systems based on our algorithms have steady-state gains of 2-6 dB or more over systems that are not rate-distortion optimized. Furthermore, our systems essentially achieve the best possible performance: the operational distortion-rate function of the source at the capacity of the packet erasure channel. [ABSTRACT FROM PUBLISHER]

Synchronization attacks like random cropping and time-scale modification are very challenging problems to audio watermarking techniques. To combat these attacks, a novel content-dependent localized robust audio watermarking scheme is proposed. The basic idea is to first select steady high-energy local regions that represent music edges like note attacks, transitions or drum sounds by using different methods, then embed the watermark in these regions. Such regions are of great importance to the understanding of music and will not be changed much for maintaining high auditory quality. In this way, the embedded watermark has the potential to escape all kinds of distortions. Experimental results show strong robustness against common audio signal processing, time-domain synchronization attacks, and most distortions introduced in Stirmark for Audio. [ABSTRACT FROM PUBLISHER]

This work proposes a method of embedding digital watermarks into audio signals in the time domain. The proposed algorithm exploits differential average-of-absolute-amplitude relations within each group of audio samples to represent one-bit information. The principle of low-frequency amplitude modification is employed to scale amplitudes in a group manner (unlike the sample-by-sample manner as used in pseudonoise or spread-spectrum techniques) in selected sections of samples so that the time-domain waveform envelope can be almost preserved. Besides, when the frequency-domain characteristics of the watermark signal are controlled by applying absolute hearing thresholds in the psychoacoustic model, the distortion associated with watermarking is hardly perceivable by human ears. The watermark can be blindly extracted without knowledge of the original signal. Subjective and objective tests reveal that the proposed watermarking scheme maintains high audio quality and is simultaneously highly robust to pirate attacks, including MP3 compression, low-pass filtering, amplitude scaling, time scaling, digital-to-analog/analog-to-digital reacquisition, cropping, sampling rate change, and bit resolution transformation. Security of embedded watermarks is enhanced by adopting unequal section lengths determined by a secret key. [ABSTRACT FROM PUBLISHER]

Two approaches for integrating encryption with multimedia compression systems are studied in this research, i.e., selective encryption and modified entropy coders with multiple statistical models. First, we examine the limitations of selective encryption using cryptanalysis, and provide examples that use selective encryption successfully. Two rules to determine whether selective encryption is suitable for a compression system are concluded. Next, we propose another approach that turns entropy coders into encryption ciphers using multiple statistical models. Two specific encryption schemes are obtained by applying this approach to the Huffman coder and the QM coder. It is shown that security is achieved without sacrificing the compression performance and the computational speed. This modified entropy coding methodology can be applied to most modern compressed audio/video such as MPEG audio, MPEG video, and JPEG/JPEG2000 images. [ABSTRACT FROM PUBLISHER]

Xiong and Malvar recently introduced a nonuniform modulated complex lapped transform (NMCLT) with good time-localization and controllable frequency resolution by using an oversampled nonuniform filter bank to generate its real and the imaginary components. In this paper, we first show that oversampling in the NMCLT is not necessary in theory but a by-product of fast implementation in practice. We also point out that the amount of oversampling, which can be flexibly controlled, depends on the application. We then describe in detail the implementation of the inverse transform, which was not addressed clearly by Xiong and Malvar. We present the first applications of the NMCLT to audio coding and image denoising. A scalable audio coder has been implemented by controlling the amount of oversampling and exploiting redundancy among the NMCLT coefficients via predictive coding. Experimental results show that the audio coder reduces pre-echoes and improves the sound quality of audio clips with transient sounds. A simple denoising algorithm based on the NMCLT has also been devised to provide images with better visual quality than those obtained with wavelet-based soft thresholding. [ABSTRACT FROM PUBLISHER]

In most watermarking systems, masking models, inherited from data compression algorithms, are used to preserve fidelity by controlling the perceived distortion resulting from adding the watermark to the original signal. So far, little attention has been paid to the consequences of using such models on a key design parameter: the robustness of the watermark to intentional attacks. The goal of this paper is to demonstrate that by considering fidelity alone, key information on the location and strength of the watermark may become available to an attacker; the latter can exploit such knowledge to build an effective mask attack. First, defining a theoretical framework in which analytical expressions for masking and watermarking are laid, a relation between the decrease of the detection statistic and the introduced perceptual distortion is found for the mask attack. The latter is compared to the Wiener filter attack. Then, considering masking models widely used in watermarking, experiments on both simulated and real data (audio and images) demonstrate how knowledge on the mask enables to greatly reduce the detection statistic, even for small perceptual distortion costs. The critical tradeoff between robustness and distortion is further discussed, and conclusions on the use of masking models in watermarking drawn. [ABSTRACT FROM PUBLISHER]

Future audio applications will need to handle a wide variety of end-devices and networks from low quality mobile services to very high quality home entertainment services. The MPEG-21 Digital Item Adaptation standard specifies a variety of tools that aim to enhance the user experience and maximize the perceived quality of service. This paper focuses on the audio-related tools, including audio usage environment descriptions and techniques for audio resource adaptation. Additionally, subjective and objective measures are discussed in the context of bit-rate adaptation. New quality measures, brightness, cleanness, and wideness are proposed for controlling the quality of the audio transmission. [ABSTRACT FROM PUBLISHER]

Conventional watermarking techniques based on echo hiding provide many benefits, but also have several disadvantages, for example, a lenient decoding process, weakness against multiple encoding attacks, etc. In this paper, to improve the weak points of conventional echo hiding, we propose a time-spread echo as an alternative to the single echo in conventional echo hiding. Spreading an echo in the time domain is achieved by using pseudonoise (PN) sequences. By spreading the echo, the amplitude of each echo can be reduced, i.e., the energy of each echo becomes small, so that the distortion induced by watermarking is imperceptible to humans while the decoding performance of the embedded watermarks is better maintained as compared with the case of conventional echo hiding, as shown by computer simulations, in which several parameters, such as the amplitude and length of PN sequences and analysis window length, were varied. Robustness against typical signal processing was also evaluated in these simulations and showed fair performance. Results of a listening test using some pieces of music showed good imperceptibility. [ABSTRACT FROM PUBLISHER]

Streaming high-fidelity audio over wireless Internet protocol (IP) networks is a challenging task because the networks present not only packet losses, but also residual bit errors. These losses and errors have severe adverse effect on the compressed audio bitstream. To solve this problem, this paper introduces error resilience in conjunction with error protection for scalable audio streaming over wireless networks. Specifically, error resilience is achieved by performing bitstream data partitioning and reversible variable length coding in the audio coder. Error protection is provided by layered product channel code to simultaneously handle packet losses and residual bit errors. Both the row and column codes of the product code provide unequal error protection for different layers of the audio bitstream by considering the characteristics of the scalable audio. Rate-distortion optimization is performed to determine the best source-channel coding tradeoff that minimizes the average expected end-to-end distortion. Simulation results demonstrate the effectiveness of our proposed approach. [ABSTRACT FROM PUBLISHER]

We present the virtual acoustics modeling framework that is a part of the MPEG-4 standard. A scene description language called the binary format for scenes (BIFS) is defined within MPEG-4 for making multimedia presentations that include various types of audio and visual data. BIFS also provides means for creating three-dimensional (3-D) virtual worlds or scenes, where visual and sound objects can be positioned and given temporal behavior. Local interaction between the user and the scene can be added to MPEG-4 applications. Typically the user can navigate in a 3-D scene so that it is viewed from different positions. In case that there are sound source objects in a scene, the sounds may be spatialized so that they are heard coming from the positions defined for them. A subset of BIFS, called the advanced AudioBIFS aims at enhanced modeling of 3-D sound environments. In this framework, sounds can be given positions, and also the virtual environment where they appear can be associated with acoustic properties that allow modeling of phenomena such as air absorption, Doppler effect, sound reflections, and reverberation. These features can be used for adding room acoustic effects to sound in the MPEG-4 terminal, and for creating immersive 3-D audiovisual scenes. [ABSTRACT FROM PUBLISHER]

The audio watermarking method proposed in this paper offers copyright protection to an audio signal by time domain processing. The strength of audio signal modifications is limited by the necessity to produce an output signal that is perceptually similar to the original one. The watermarking method presented here does not require the use of the original signal for watermark detection. The watermark signal is generated using a key, i.e., a single number known only to the copyright owner. Watermark embedding depends on the audio signal amplitude and frequency in a way that minimizes the audibility of the watermark signal. The embedded watermark is robust to common audio signal manipulations like MPEG audio coding, cropping, time shifting, filtering, resampling, and requantization [ABSTRACT FROM PUBLISHER]

We present an overview of the AudioBIFS system, part of the Binary Format for Scene Description (BIFS) tool in the MPEG-4 International Standard. AudioBIFS is the tool that integrates the synthetic and natural sound coding functions in MPEG-4. It allows the flexible construction of soundtracks and sound scenes using compressed sound, sound synthesis, streaming audio, interactive and terminal-dependent presentation, three-dimensional (3-D) spatialization, environmental auralization, and dynamic download of custom signal-processing effects algorithms. MPEG-4 sound scenes are based on a model that is a superset of the model in VRML 2.0, and we describe how MPEG-4 is built upon VRML and the new capabilities provided by MPEG-4. We discuss the use of structured audio orchestra language, the MPEG-4 SAOL, for writing downloadable effects, present an example sound scene built with AudioBIFS, and describe the current state of implementations of the standard [ABSTRACT FROM PUBLISHER]