Complexity-scalable bit detection with MP3 audio bitstreams

Master'sMASTER OF SCIENC

A new data embedding method for mpeg layer III audio steganography

A new method of MP3 steganography is proposed with emphasis on increasing the steganography capacity of the carrier medium. This paper proposes a data embedding algorithm to hide more information for compressed bitstream of MP3 audio files. The sign bits of Huffman codes are selected as the stego-object according to the Huffman coding characteristic in region of Count1. Embedding process does not require the main MP3 audio file during the extraction of hidden message and the size of MP3 file cannot be changed in this step. Our proposed method caused much higher information embedding capacity with lower computational complexity compared with MP3Stego tools. Experimental results show an excellent imperceptibility for the new algorithm

Electrical Network Frequency as a Tool for Audio Concealment Process

[[abstract]]We live in a digital era. Digital contents may be produced by digital equipments or by converting old analog recordings. With the rapid growth of digital contents, digital archiving technology is demanded. Different types of contents require different processing techniques. In this paper, we focus on digital audio contents. The related techniques, such as forensics, authentication, and error concealment, were studied. When converting audio tapes to digital files, sometimes a certain automatic error detection and concealment is needed. However, traditional audio tapes were recorded without any error recovery information. Based on the restriction, we proposed a scheme that incorporates the electrical network frequency (ENF) as a tool for detecting damaged audio segments. The goal is to help people identifying candidate concealment segments. When using in an archiving application, it reduces the manpower as well as increases the accuracy of the generated meta-data.[[conferencetype]]國際[[conferencedate]]20101015~20101017[[iscallforpapers]]Y[[conferencelocation]]Darmstadt, German

Alignment of the recording of a violin performance with the corresponding musical score

Català: En aquest projecte es pretén estudiar una part de la conducta dels músics, com és la variació del tempo al llarg de la cançó. Per fer-ho, s'ha d'obtenir la durada de les notes durant la interpretació de les peces. Les notes tenen una durada determinada a la partitura, però aquesta varia substancialment respecte la durada de les notes tocades pels músics, ja que aquests toquen sense l'ajut de metrònom. És per això que existeix una variació en el tempo de la cançó respecte el tempo teòric. Obtenint la durada real de les notes, es pot calcular el tempo seguit pel músic. En el nostre cas els músics estudiats són violinistes. El procediment per analitzar els temes musicals enregistrats consisteix en la gravació del so, i en la utilització d'uns sensors per obtenir la posició, el moviment, la força i la velocitat de l'arc durant la peça. Amb tots aquests paràmetres és possible aconseguir un model definit de la interpretació del músic. Per calcular el tempo s?ha implementat un algorisme, anomenat detector de polsos o beat detector, que detecta el tempo seguit pel violinista durant la interpretació. Posteriorment, s?ha implementat un altre algorisme, anomenat alineador o aligner que, aprofitant els resultats de la variació del tempo, crea una partitura amb les duracions reals de cada nota. Actualment, els sistemes existents que detecten el tempo de les cançons utilitzen només les dades de la gravació del so. Per tal de millorar l'eficiència d'aquests algorismes, proposem un nou sistema que, a part d'utilitzar el so, incorpora les dades provinents del moviment de l'arc. Finalment, per tal de millorar els resultats obtinguts, s'ha proposat un sistema que detecta els polsos de les cançons utilitzant les dades provinents d'un duet, trio o quartet com a senyals d'entrada. El fet que hi hagi més d'una veu en una cançó implica que hi hagi una relació implícita entre els tempos que segueixen els violinistes, que és la que utilitza l'algorisme proposat.Castellano: En este proyecto se pretende estudiar una parte de la conducta de los músicos, como es la variación del tempo a lo largo de la canción. Para hacerlo, se obtiene la duración de las notas durante la interpretación de las piezas. Las notas tienen una duración determinada en la partitura, pero ésta varia sustancialmente respecto la duración de las notas tocadas por los músicos, ya que éstos tocan sin la ayuda de metrónomo. Es por eso que existe una variación en el tempo de la canción respecto al tempo teórico. Obteniendo la duración real de las notas, se puede calcular el tempo seguido por el músico. En nuestro caso los músicos estudiados son violinistas. El procedimiento para analizar los temas musicales grabados consiste en la grabación del sonido y en la utilización de unos sensores para obtener la posición, el movimiento, la fuerza y la velocidad del arco durante la pieza. Con todos estos parámetros es posible conseguir un modelo definido de la interpretación del músico. Actualmente, los sistemas existentes que detectan el tempo de las canciones utilizan sólo los datos de la grabación del sonido. Con tal de mejorar la eficiencia de estos algoritmos, proponemos un nuevo sistema que, a parte de utilizar el sonido, incorpora los datos provenientes del movimiento del arco. Finalmente, con tal de mejorar los resultados obtenidos, se propone un sistema que detecta los pulsos de las canciones utilizando los datos provenientes de un dueto, trío o cuarteto como señales de entrada. El hecho de que haya más de una voz en una canción implica que haya una relación implícita entres los tempos que siguen los violinistas, que es la que utiliza el algoritmo propuesto.English: This work aims to study some of the behavior of the musicians regarding the change of the tempo throughout the song. In order to do it, the length of the notes during the performance of the pieces has to be computed. Although the notes have a fixed length marked in the score, it varies substantially with the duration of the notes of the song. Because they play without the help of a metronome, this is why there is a variation of the tempo of the songs compared with the theoretical one. The actual tempo followed by the musicians can be computed by calculating the actual duration of the notes. In our case, the musicians studied are violinists. The procedure to analyze the recorded songs consists of using some sensors to obtain the position, the velocity, the movement and the force of the bow used in the performance. With all these parameters it is possible to define an accurate model of the musical performance. Currently, the existing systems that detect the tempo of the songs only use the recording data as an input. In order to improve the efficiency of these algorithms, we propose a new system that not only uses the recording, but also the data of the bow displacement. Finally, in order to improve the obtained results, we propose another system that detects the beats of the songs using the data of a duet, trio or quartet as inputs. In these cases, there is more than one violin playing at the same time, and there is an implicit relation between the tempos followed by the violinists. The proposed algorithm takes advantage of this relation to obtain the actual tempo

Efficiency in audio processing : filter banks and transcoding

Audio transcoding is the conversion of digital audio from one compressed form A to another compressed form B, where A and B have different compression properties, such as a different bit-rate, sampling frequency or compression method. This is typically achieved by decoding A to an intermediate uncompressed form, and then encoding it to B. A significant portion of the involved computational effort pertains to operating the synthesis filter bank, which is an important processing block in the decoding stage, and the analysis filter bank, which is an important processing block in the encoding stage. This thesis presents methods for efficient implementations of filter banks and audio transcoders, and is separated into two main parts. In the first part, a new class of Frequency Response Masking (FRM) filter banks is introduced. These filter banks are usually characterized by comprising a tree-structured cascade of subfilters, which have small individual filter lengths. Methods of complexity reduction are proposed for the scenarios when the filter banks are operated in single-rate mode, and when they are operated in multirate mode; and for the scenarios when the input signal is real-valued, and when it is complex-valued. An efficient variable bandwidth FRM filter bank is designed by using signed-powers-of-two reduction of its subfilter coefficients. Our design has a complexity an order lower than that of an octave filter bank with the same specifications. In the second part, the audio transcoding process is analyzed. Audio transcoding is modeled as a cascaded quantization process, and the cascaded quantization of an input signal is analyzed under different conditions, for the MPEG 1 Layer 2 and MP3 compression methods. One condition is the input-to-output delay of the transcoder, which is known to have an impact on the audio quality of the transcoded material. Methods to reduce the error in a cascaded quantization process are also proposed. An ultra-fast MP3 transcoder that requires only integer operations is proposed and implemented in software. Our implementation shows an improvement by a factor of 5 to 16 over other best known transcoders in terms of execution speed

Alignment of the recording of a violin performance with the corresponding musical score

Català: En aquest projecte es pretén estudiar una part de la conducta dels músics, com és la variació del tempo al llarg de la cançó. Per fer-ho, s'ha d'obtenir la durada de les notes durant la interpretació de les peces. Les notes tenen una durada determinada a la partitura, però aquesta varia substancialment respecte la durada de les notes tocades pels músics, ja que aquests toquen sense l'ajut de metrònom. És per això que existeix una variació en el tempo de la cançó respecte el tempo teòric. Obtenint la durada real de les notes, es pot calcular el tempo seguit pel músic. En el nostre cas els músics estudiats són violinistes. El procediment per analitzar els temes musicals enregistrats consisteix en la gravació del so, i en la utilització d'uns sensors per obtenir la posició, el moviment, la força i la velocitat de l'arc durant la peça. Amb tots aquests paràmetres és possible aconseguir un model definit de la interpretació del músic. Per calcular el tempo s?ha implementat un algorisme, anomenat detector de polsos o beat detector, que detecta el tempo seguit pel violinista durant la interpretació. Posteriorment, s?ha implementat un altre algorisme, anomenat alineador o aligner que, aprofitant els resultats de la variació del tempo, crea una partitura amb les duracions reals de cada nota. Actualment, els sistemes existents que detecten el tempo de les cançons utilitzen només les dades de la gravació del so. Per tal de millorar l'eficiència d'aquests algorismes, proposem un nou sistema que, a part d'utilitzar el so, incorpora les dades provinents del moviment de l'arc. Finalment, per tal de millorar els resultats obtinguts, s'ha proposat un sistema que detecta els polsos de les cançons utilitzant les dades provinents d'un duet, trio o quartet com a senyals d'entrada. El fet que hi hagi més d'una veu en una cançó implica que hi hagi una relació implícita entre els tempos que segueixen els violinistes, que és la que utilitza l'algorisme proposat.Castellano: En este proyecto se pretende estudiar una parte de la conducta de los músicos, como es la variación del tempo a lo largo de la canción. Para hacerlo, se obtiene la duración de las notas durante la interpretación de las piezas. Las notas tienen una duración determinada en la partitura, pero ésta varia sustancialmente respecto la duración de las notas tocadas por los músicos, ya que éstos tocan sin la ayuda de metrónomo. Es por eso que existe una variación en el tempo de la canción respecto al tempo teórico. Obteniendo la duración real de las notas, se puede calcular el tempo seguido por el músico. En nuestro caso los músicos estudiados son violinistas. El procedimiento para analizar los temas musicales grabados consiste en la grabación del sonido y en la utilización de unos sensores para obtener la posición, el movimiento, la fuerza y la velocidad del arco durante la pieza. Con todos estos parámetros es posible conseguir un modelo definido de la interpretación del músico. Actualmente, los sistemas existentes que detectan el tempo de las canciones utilizan sólo los datos de la grabación del sonido. Con tal de mejorar la eficiencia de estos algoritmos, proponemos un nuevo sistema que, a parte de utilizar el sonido, incorpora los datos provenientes del movimiento del arco. Finalmente, con tal de mejorar los resultados obtenidos, se propone un sistema que detecta los pulsos de las canciones utilizando los datos provenientes de un dueto, trío o cuarteto como señales de entrada. El hecho de que haya más de una voz en una canción implica que haya una relación implícita entres los tempos que siguen los violinistas, que es la que utiliza el algoritmo propuesto.English: This work aims to study some of the behavior of the musicians regarding the change of the tempo throughout the song. In order to do it, the length of the notes during the performance of the pieces has to be computed. Although the notes have a fixed length marked in the score, it varies substantially with the duration of the notes of the song. Because they play without the help of a metronome, this is why there is a variation of the tempo of the songs compared with the theoretical one. The actual tempo followed by the musicians can be computed by calculating the actual duration of the notes. In our case, the musicians studied are violinists. The procedure to analyze the recorded songs consists of using some sensors to obtain the position, the velocity, the movement and the force of the bow used in the performance. With all these parameters it is possible to define an accurate model of the musical performance. Currently, the existing systems that detect the tempo of the songs only use the recording data as an input. In order to improve the efficiency of these algorithms, we propose a new system that not only uses the recording, but also the data of the bow displacement. Finally, in order to improve the obtained results, we propose another system that detects the beats of the songs using the data of a duet, trio or quartet as inputs. In these cases, there is more than one violin playing at the same time, and there is an implicit relation between the tempos followed by the violinists. The proposed algorithm takes advantage of this relation to obtain the actual tempo

Digital Watermarking for Verification of Perception-based Integrity of Audio Data

In certain application fields digital audio recordings contain sensitive content. Examples are historical archival material in public archives that preserve our cultural heritage, or digital evidence in the context of law enforcement and civil proceedings. Because of the powerful capabilities of modern editing tools for multimedia such material is vulnerable to doctoring of the content and forgery of its origin with malicious intent. Also inadvertent data modification and mistaken origin can be caused by human error. Hence, the credibility and provenience in terms of an unadulterated and genuine state of such audio content and the confidence about its origin are critical factors. To address this issue, this PhD thesis proposes a mechanism for verifying the integrity and authenticity of digital sound recordings. It is designed and implemented to be insensitive to common post-processing operations of the audio data that influence the subjective acoustic perception only marginally (if at all). Examples of such operations include lossy compression that maintains a high sound quality of the audio media, or lossless format conversions. It is the objective to avoid de facto false alarms that would be expectedly observable in standard crypto-based authentication protocols in the presence of these legitimate post-processing. For achieving this, a feasible combination of the techniques of digital watermarking and audio-specific hashing is investigated. At first, a suitable secret-key dependent audio hashing algorithm is developed. It incorporates and enhances so-called audio fingerprinting technology from the state of the art in contentbased audio identification. The presented algorithm (denoted as ”rMAC” message authentication code) allows ”perception-based” verification of integrity. This means classifying integrity breaches as such not before they become audible. As another objective, this rMAC is embedded and stored silently inside the audio media by means of audio watermarking technology. This approach allows maintaining the authentication code across the above-mentioned admissible post-processing operations and making it available for integrity verification at a later date. For this, an existent secret-key ependent audio watermarking algorithm is used and enhanced in this thesis work. To some extent, the dependency of the rMAC and of the watermarking processing from a secret key also allows authenticating the origin of a protected audio. To elaborate on this security aspect, this work also estimates the brute-force efforts of an adversary attacking this combined rMAC-watermarking approach. The experimental results show that the proposed method provides a good distinction and classification performance of authentic versus doctored audio content. It also allows the temporal localization of audible data modification within a protected audio file. The experimental evaluation finally provides recommendations about technical configuration settings of the combined watermarking-hashing approach. Beyond the main topic of perception-based data integrity and data authenticity for audio, this PhD work provides new general findings in the fields of audio fingerprinting and digital watermarking. The main contributions of this PhD were published and presented mainly at conferences about multimedia security. These publications were cited by a number of other authors and hence had some impact on their works

Content-based music structure analysis

Ph.DDOCTOR OF PHILOSOPH

Multimedia

The nowadays ubiquitous and effortless digital data capture and processing capabilities offered by the majority of devices, lead to an unprecedented penetration of multimedia content in our everyday life. To make the most of this phenomenon, the rapidly increasing volume and usage of digitised content requires constant re-evaluation and adaptation of multimedia methodologies, in order to meet the relentless change of requirements from both the user and system perspectives. Advances in Multimedia provides readers with an overview of the ever-growing field of multimedia by bringing together various research studies and surveys from different subfields that point out such important aspects. Some of the main topics that this book deals with include: multimedia management in peer-to-peer structures & wireless networks, security characteristics in multimedia, semantic gap bridging for multimedia content and novel multimedia applications