Audio Bit Research Articles

A Modified Enhanced Method of Audio – Video Steganography for High Security Data Transmission

Steganography is an art of convert communication, which offers secrete and secure way of communication. It has many application areas such as audio-video synchronization,in defense forces etc… The audio data is converted into binary format and encrypted using a XOR encryption algorithm. Then the encrypted audio data converted into XL data form. The video frames are divided into non-overlapping blocks, and the difference between the adjacent pixels in each block is computed to obtain a reference matrix. Each block is then iterated through, if the difference between adjacent pixels is greater than or equal to the average difference in the reference matrix, the MPVD (Modified Pixel Value Differencing) based LSB of the pixel value is modified to embed the encrypted audio data bit. The embedded bit is then XORed with the corresponding LSB of adjacent pixel in the same block. The modified video frames can be sent over an unsecured channel without revealing the embedded encrypted audio data. To extract the encrypted audio message from the stego video, the embedded bits are XORed with the corresponding LSB of the adjacent pixel in the same block, and the modified LSBs are extracted and decrypted using the same encryption algorithm and key.

A subjective and objective evaluation of a codec for the electrical stimulation patterns of cochlear implants.

Wireless transmission of audio from or to signal processors of cochlear implants (CIs) is used to improve speech understanding of CI users. This transmission requires wireless communication to exchange the necessary data. Because they are battery powered devices, energy consumption needs to be kept low in CIs, therefore making bitrate reduction of the audio signals necessary. Additionally, low latency is essential. Previously, a codec for the electrodograms of CIs, called the Electrocodec, was proposed. In this work, a subjective evaluation of the Electrocodec is presented, which investigates the impact of the codec on monaural speech performance. The Electrocodec is evaluated with respect to speech recognition and quality in ten CI users and compared to the Opus audio codec. Opus is a low latency and low bitrate audio codec that best met the CI requirements in terms of bandwidth, bitrate, and latency. Achieving equal speech recognition and quality as Opus, the Electrocodec achieves lower mean bitrates than Opus. Actual rates vary from 24.3 up to 53.5 kbit/s, depending on the codec settings. While Opus has a minimum algorithmic latency of 5 ms, the Electrocodec has an algorithmic latency of 0 ms.

Maturity classification of cacao through spectrogram and convolutional neural network

Cacao pod's ideal harvesting time is when it is about to be ripe. Immature harvest would result in hard cacao beans not suitable for fermentation, while overripe cacao pods lead to fungal-infected, defective, and poor-quality yields. The demand for high-quality cacao products is expected to rise due to advancing technology in the present. Pre-harvesting needs to provide optimal identification of which amongst the pods are ripened enough and ready for the next stage of the cacao process. This paper recommends a technique to determine the ripeness of cacao. Nine hundred thirty-three cacao samples were used to collect thumping audio data at five different pod's exocarp locations. Each sound file is 1 second long, creating 4665 cacao sound file datasets at 16kHz sample rate and 16-bit audio bit depth. The process of the Mel-Frequency Cepstral Coefficient Spectogram was then applied to extract recognizable features for the training process. The deep learning method integrated was a convolutional neural network (CNN) to classify the cacao sound successfully. The experimental design model's output exhibits an accuracy of 97.50 % for the training data and 97.13 % for the validation data. While the overall accuracy mean of the classification system is 97.46 %, whether the cacao is unripe or ripe.

Audiovisual quality of live music streaming over mobile networks using MPEG-DASH

The MPEG-DASH protocol has been rapidly adopted by most major network content providers and enables clients to make informed decisions in the context of HTTP streaming, based on network and device conditions using the available media representations. A review of the literature on adaptive streaming over mobile shows that most emphasis has been on adapting the video quality whereas this work examines the trade-off between video and audio quality. In particular, subjective tests were undertaken for live music streaming over emulated mobile networks with MPEG-DASH. A group of audio/video sequences was designed to emulate varying bandwidth arising from network congestion, with varying trade-off between audio and video bit rates. Absolute Category Rating was used to evaluate the relative impact of both audio and video quality in the overall Quality of Experience (QoE). One key finding from the statistical analysis of Mean Opinion Scores (MOS) results using Analysis of Variance indicates that providing reduced audio quality has a much lower impact on QoE than reducing video quality at similar total bandwidth situations. This paper also describes an objective model for audiovisual quality estimation that combines the outcomes from audio and video metrics into a joint parametric model. The correlation between predicted and subjective MOS was computed using several outcomes (Pearson and Spearman correlation coefficients, Root Mean Square Error (RMSE) and epsilon-insensitive RMSE). The obtained results indicate that the proposed approach is a viable solution for objective audiovisual quality assessment in the context of live music streaming over mobile network.

Autocorrelation Modulation-Based Audio Blind Watermarking Robust Against High Efficiency Advanced Audio Coding

High Efficiency Advanced Audio Coding (HE-AAC) is a lossy compression method for digital audio data which supplies high-quality audio at a very low bit rate. In this paper, the audio blind watermarking algorithm, on the basis of autocorrelation modulation, is introduced to maximize the robustness against low-bit rate HE-AAC. The watermark is embedded by modulating the normalized correlation of the original signal as well as its delayed version. The signal-to-noise ratio of before and after HE-AAC compression decides the strength of the embedding watermark. The watermarking embedding strength is guaranteed by the feedback process. The effectiveness of the proposed method is proven using the Perceptual Evaluation of Audio Quality algorithm and bit error rate of recovered watermarks under HE-AAC compression on mono, stereo and 5.1 channel audio. Experimental results show that the proposed method provides good performance in terms of imperceptibility, robustness and data payload compared with some recent state-of-the-art watermarking methods under an MPEG-2 Audio Layer III (MP3) compression attack.

Encrypted Image Watermark in Audio Files Using Homogenous Deffie-Hellman with Chebyshev Polynomial

Due to the expanding utilization of advanced media, the assurance of protected innovation rights issue has turned into an essential issue. Digital watermarking is currently drawing consideration as another technique for shielding media content from unapproved duplicating. In this paper, watermarking image (logo) will be encrypted using a key constructed by a proposed homogenous method of Diffie-Hellman and chebyshev polynomial. The encrypted watermark will be embedded in different samples from the transformed file (DCT) of audio. The embedding process will depend on binary similarity between the audio and watermark bits which reduces the effect of embedded data. The proposed method for the key generation is more secure and complicated since it combines the strength factors of both Diffie-Hellman and Chebyshev polynomial. The effect of the embedding is nonperceptibile and nondetectable.

ViSQOLAudio: An objective audio quality metric for low bitrate codecs.

Streaming services seek to optimise their use of bandwidth across audio and visual channels to maximise the quality of experience for users. This letter evaluates whether objective quality metrics can predict the audio quality for music encoded at low bitrates by comparing objective predictions with results from listener tests. Three objective metrics were benchmarked: PEAQ, POLQA, and VISQOLAudio. The results demonstrate objective metrics designed for speech quality assessment have a strong potential for quality assessment of low bitrate audio codecs.

Full-reference audio-visual video quality metric

The goal of this work is to present a full-reference (FR) audio-visual quality metric. We performed three psychophysical experiments in order to obtain a better understanding of how audio and video components interact with each other and how these interactions affect the overall audio-visual quality. In experiment I, subjects evaluated the quality of videos (without any audio) compressed at different video bitrates. In experiment II, subjects evaluated the quality of audio (without any video) compressed at different audio bitrates. In experiment III, subjects evaluated the quality of videos (audio-visual signals), which had their audio and video components compressed at different bitrates. Based on the data gathered from these experiments, we obtain a set of subjective models for audio-visual quality. Inspired by these subjective models, we propose a set of FR audio-visual quality metrics composed of a combination of a video quality metric and an audio quality metric. The proposed metrics have good performance and present better results when compared to simple FR video quality metrics.

Sinusoidal Coding and Spectral Band Replication for Low Bit-Rate Super-Wideband Speech and Audio Coding

In this paper, we present a new sinusoidal coding and spectral band replication (SBR) method for a low bit-rate super-wideband speech and audio coding. The sinusoidal coding algorithm utilizes the local rms energy of partial frequency bands in modified discrete cosine transform (MDCT) domain to select peak MDCTs. The SBR is based on the pseudo-spectral correlation between lower band and higher band. The proposed techniques are shown experimentally to give improved performance compared to ITU-T G.718 Annex B from the objective and subjective evaluation.

Parametric Audio Coding With Exponentially Damped Sinusoids

Sinusoidal modeling is one of the most popular techniques for low bitrate audio coding. Usually, the sinusoidal parameters (amplitude, pulsation and phase of each sinusoidal component) are kept constant within a time segment. An alternative model, the so-called Exponentially-Damped Sinusoidal (EDS) model, includes an additional damping parameter for each sinusoidal component to better represent the signal characteristics. It was however never shown that the EDS model could be efficient for perceptual audio coding. To that aim, we propose in this paper an efficient analysis/synthesis framework with dynamic time-segmentation on transients and psychoacoustic modeling, and an asymptotically optimal entropy-constrained quantization method for the four sinusoid parameters (e.g., including damping). We then apply this coding technique to real audio excerpts for a given entropy target corresponding to a low bitrate (20 kbits/s), and compare this method with a classical sinusoidal coding scheme using a constant-amplitude sinusoidal model and the perceptually weighted Matching Pursuit algorithm. Subjective listening tests show that the EDS model is more efficient on audio samples with fast transient content, and similar to the classical model for more stationary audio samples.

Offloading of media transcoding for high-quality multimedia services

The demand for high-quality multimedia service in handheld devices has increased with the advance of consumer electronics technologies. However, there are inherent performance limits in such mobile devices due to low processing power, requirements for low power consumption, and limited storage capacity. These factors have obviously prevented high-performance multimedia services from being feasible. In addition, most of the mobile devices support distinct multimedia data formats for video and audio codecs, bit rate, and screen size. Therefore, streaming media or receiving data from other users' devices essentially requires additional data translation along with a transcoding operation. However, most of the processors that are used in mobile devices or even in personal media servers cannot provide sufficient processing power for transcoding. To address the problem, this paper proposes an efficient offloading method for media transcoding can be applied in current commercial products. The system is designed on top of Java and provides sufficient transcoding performance for media streaming with low-end processors.

An 802.11n Wireless Local Area Network Transmission Scheme for Wireless Telemedicine Applications

In this paper, an 802.11 n transmission scheme is proposed for wireless telemedicine applications. IEEE 802.11n standards, a power assignment strategy, space-time block coding (STBC), and an object composition Petri net (OCPN) model are adopted. With the proposed wireless system, G.729 audio bit streams, Joint Photographic Experts Group 2000 (JPEG 2000) clinical images, and Moving Picture Experts Group 4 (MPEG-4) video bit streams achieve a transmission bit error rate (BER) of 10-7, 10-4, and 103 simultaneously. The proposed system meets the requirements prescribed for wireless telemedicine applications. An essential feature of this proposed transmission scheme is that clinical information that requires a high quality of service (QoS) is transmitted at a high power transmission rate with significant error protection. For maximizing resource utilization and minimizing the total transmission power, STBC and adaptive modulation techniques are used in the proposed 802.11 n wireless telemedicine system. Further, low power, direct mapping (DM), low-error protection scheme, and high-level modulation are adopted for messages that can tolerate a high BER. With the proposed transmission scheme, the required reliability of communication can be achieved. Our simulation results have shown that the proposed 802.11 n transmission scheme can be used for developing effective wireless telemedicine systems.

Parametric Packet-Layer Model for Evaluation Audio Quality in Multimedia Streaming Services

We propose a parametric packet-layer model for monitoring audio quality in multimedia streaming services such as Internet protocol television (IPTV). This model estimates audio quality of experience (QoE) on the basis of quality degradation due to coding and packet loss of an audio sequence. The input parameters of this model are audio bit rate, sampling rate, frame length, packet-loss frequency, and average burst length. Audio bit rate, packet-loss frequency, and average burst length are calculated from header information in received IP packets. For sampling rate, frame length, and audio codec type, the values or the names used in monitored services are input into this model directly. We performed a subjective listening test to examine the relationships between these input parameters and perceived audio quality. The codec used in this test was the Advanced Audio Codec-Low Complexity (AAC-LC), which is one of the international standards for audio coding. On the basis of the test results, we developed an audio quality evaluation model. The verification results indicate that audio quality estimated by the proposed model has a high correlation with perceived audio quality.

“AQUA”: A New Audio Quality Monitoring Algorithm for the Performance of DRM Receivers in Fading Channels

This paper presents a new audio quality monitoring algorithm: ldquoAQUArdquo for assessing the performance of Digital Radio Mondiale (DRM) receivers. AQUA targets performance evaluation of consumer grade receivers in fading channels objectively. Consumer grade DRM receivers do not generally possess a Receiver Status and Control Interface (RSCI), which prohibits the possibility of analysing digitally decoded audio bit stream after reception. AQUA relies on the analysis of the received analog audio output signal instead. The paper describes the development of the AQUA algorithm and the measurement system used for receiver audio quality testing. The performance of AQUA is compared to conventional RSCI derived metrics like Bit Error Rate (BER) and Frame Error Rate (FER) of the decoded audio data. A series of tests are conducted in a variety of propagation conditions using a DRM dedicated testing system with a fading channel emulator. It is shown that FER estimates using AQUA are in good agreement with conventional FER figures derived from RSCI in different fading channel scenarios. This allows AQUA to be used as a reliable alternative to conventional FER performance analysis for automated objective testing of DRM receivers.

On the Study of Noise Allocation for Speech Signal in Low Bit-Rate Audio Coding

This letter proposes a new masking threshold adjustment method to improve the quality for the speech signals in low bit-rate audio coding. The Enhanced aacPlus (EAAC) audio codec increases the masking threshold of all frequency bands to be suitable for the given encoding rate by considering equal loudness noises only, which is a representative way for implementing the adjustment technique. The proposed method, however, dynamically adjusts the masking threshold of each frequency band based on the energy ratio of each band to the average band energy. More quantization noises are added to formant regions that have relatively large energy ratio values, but less distortion is allowed in spectral valley regions, which eventually helps to enhance perceptual quality for speech signals. The proposed idea reflects the spectral weighting criterion in searching optimal excitation codebooks used in many speech coding algorithms. Simulation results confirm that the proposed method implemented on the EAAC coder improves quality for the speech input signals at the same bit-rate while keeping equivalent quality for music contents.

Multiplexing H.264 video with AAC audio bit streams

This paper has developed a multiplexing-demultiplexing system for H.264 video stream and AAC audio stream that is based on theMPEG-2 framework and uses frame numbers as timestamps. It also provides lip sync after decoding the video and audio bit streams. .

A bit reduction algorithm for Spectral Band Replication based on human auditory characteristics

Spectral band replication (SBR) is an effective tool for low bitrate audio codecs such as HE-AAC and enhanced aacPlus. We propose a new algorithm for the bitrate reduction of SBR, which modifies the envelope data coding of SBR considering the threshold in quiet and the masking threshold. Experimental results show that the proposed algorithm significantly reduces the bitrate compared to the conventional SBR with no degradation in perceived sound quality.

This is not a C#: What My Computer Means

AbstractComputers contain layers of abstraction between the raw audio bits and the user experience; each layer requires a method of traversal that reflects the attitudes of the programmer/composer/performer. The navigation of this hierarchy results in modes of thought that affect notions of how musical signifiers can be subverted and redefined. This explosion of individualistic technological systems defines meaning for new sonic results, devoid of traditional communicative signs; the manipulations within these systems directly assail the traditional signification of artistic value and of institutional canonisation. A few pieces and scenarios are presented as examples. Additionally presented are two opposing tendencies (with some empirical manifestations) in the creation of live performances systems, here labelled as extensional and algorithmic. The article posits that a performance system cannot amplify human expression without necessarily having either predefined processes or predefined limitations, and that the choice in this design is reflected in the composer's attitude towards layer-traversal.

Video bookmark based on soundtrack identification and two-stage search for interactive-television

This paper presents a video retrieval system (VRS) for Interactive-Television as like internet protocol television (IPTV). A video bookmark initiated by users is performed based on snippets of the background soundtrack corresponding to the ongoing program. Our VRS has two special aspects compared with previous bookmark systems. First, we adopt the robust audio fingerprint feature of long-term logarithmic modified DCT modulation coefficients (LMDCT-MC) for audio indexing and retrieval. Second, we propose and apply a two-stage search (TSS) algorithm for fast searching. In the first stage of TSS, candidate video segments are roughly determined with audio index bit vectors (IBV) and then the optimal video clip is obtained by fingerprint bit vectors (FBV). We evaluate the proposed system with a database of 100 TV programs including news, panel discussions, music shows, advertisements, and dramas. The experimental results show that our VRS achieve fast search, robustness to noise and high precision of retrieval. A search accuracy of 99.67% was accomplished.

Mobile Video: A Study of Quality Perception

This paper presents an evaluation of the perception of streaming multimedia quality when delivered over a limited bandwidth, third generation (3G) wireless data network. The study has three primary objectives: 1) measure the perceived quality of the multimedia experience as a function of frame rate, audio fidelity and data connection speed, 2) identify where the perceived quality of the multimedia experience shifts from being acceptable to unacceptable, and 3) identify the optimal combination of frame rate, audio fidelity and connection speed that provides the best customer experience. Analysis was conducted using Response Surface Methodology (RSM). RSM was used to identify how the three parameters of frame rate, audio stream bit rate and data connection speed interact and affect significant shifts in the perceived quality of the multimedia experience. Results identify the parameters and their characteristics that are related to shifts in the perceived quality across five types of multimedia content. The findings of this study are used to guide multimedia content developers on how to create (i.e., coding and encoding) multimedia content to ensure the best possible experience over limited bandwidth 3G wireless data networks. In addition, this research has implications for the design of future mobile data networks that will use quality of service technologies to ensure the best possible customer experience.