Parametric vector quantization for coding percussive sounds in music

We propose a frame loss concealment technique for decoders compatible with MPEG advanced audio coding (AAC). The spectral information of the lost frame is estimated in the modified discrete cosine transform (MDCT) domain via efficient techniques that are tailored to individual source signal components: in noise-like spectral bins the MDCT coefficients are obtained by shaped-noise insertion, while coefficients in tone-dominant bins are estimated by frame interpolation followed by a refinement procedure so as to optimize the fit of the concealed frames with neighboring frames. Experimental results demonstrate that the proposed technique offers performance superior to techniques adopted in commercial AAC decoders.

In this paper, we develop a 3D audio reproduction scheme for the purpose of delivering audio over IP networks. In this scheme, audio streams constructed at a server are composed of the TCP/IP header followed by multi-channel audio data compressed by MPEG advanced audio coding (AAC), and the decoded audio signals are played out on a stereo loudspeaker system at the client. Since the audio source is recorded by a multi-channel microphone but the playout is dedicated to stereo speakers, the quality mismatch between the multi-channel and the stereo system should be overcome. As a potential solution, we first investigate the effect of 3D audio processing on the audio quality at the client by applying a head-related transfer function (HRTF). Next, a crosstalk cancellation process is applied to the audio with 3D effects in order to improve the immersion of the processed 3D effects on a stereo loudspeaker system. Finally, we evaluate the performance of the 3D audio reproduction system in terms of the identification of an audio source and quality comparison before and after applying the crosstalk cancellation technique.

Audio compression systems exploit periodicity in signals to remove inter-frame redundancies via the long term prediction (LTP) tool. This simple tool capitalizes on the periodic component of the waveform by selecting a past segment as the basis for prediction of the current frame. However, most audio signals are polyphonic in nature, containing a mixture of several periodic components. While such polyphonic signals may themselves be periodic with overall period equaling the least common multiple of the individual component periods, the signal rarely remains sufficiently stationary over the extended period, rendering the LTP tool suboptimal. Instead of seeking a past segment that represents a “compromise” for incompatible component periods, we propose a more complex filter that predicts every periodic component of the signal from its immediate history, and this is achieved by cascading LTP filters, each corresponding to individual periodic component. We also propose a recursive “divide and conquer” technique to estimate parameters of all the LTP filters. We then demonstrate the effectiveness of cascaded LTP in two distinct settings of the ultra low delay Bluetooth Subband Codec and the MPEG Advanced Audio Coding (AAC) standard. In MPEG AAC, we specifically adapt the cascaded LTP parameter estimation to take into account the perceptual distortion criteria, and also propose a low decoder complexity variant. Objective and subjective results for all the settings validate the effectiveness of the proposal on a variety of polyphonic signals.

We describe an efficient implementation for converting MPEG-2/MPEG-4 Advanced Audio Coding (AAC) encoded data to Dolby Digital AC-3. We describe many techniques to exploit the information in the AAC bitstream to simplify the AC-3 encoder. These techniques can be straightforwardly used in other transcoding schemes between different multimedia standards.

In this paper, we describe the implementation of the multi-channel AAC encoder system for digital audio broadcasting. The encoder system is based on MPEG-2/4 advanced audio coding (AAC) and is capable of real-time encoding up to 5.1 channel audio. To give a flexible functionality, it consists of multiple DSPs, IEC61937 and TCP/IP interface and 6 channel audio input facilities. The reference AAC decoder was implemented for verification test of the encoder. The encoder system is also integrated with the AAC streaming system for interoperation test. Through these tests, the encoder system is verified to be a good solution for high quality audio broadcasting.

Bandwidth efficiency and error robustness are essential issues for different multimedia streaming applications. This paper presents strategies for high-quality audio streaming based on fragmenting perceptually coded audio frames and shuffling the data components among multiple packets for transportation. This is done to increase robustness against packet loss. We also address the delivery of audio data consisting of components with different proportional priorities. Our approach is rationalized with streaming tests using the MPEG AAC audio codec in a simulated network environment and formal listening tests to evaluate the resulting audio output. According to the results, the proposed schemes improve audio quality significantly with reasonable increase to network resource utilization compared to traditional error robustness measures.

In this paper, the relationship between the features of the input audio and its corresponding transparent bitrate is studied. This relationship can be used to pre-estimate the transparent bitrate required for an input audio file, so that this bitrate can be used for encoding and the transparent quality can be achieved with the minimum compressed file size. Two features are selected and tested. Firstly, the perceptual feature is extracted from the test sequences using MPEG-4 advanced audio coding. The results of the signal-to-mask ratio and the transparent bitrate are compared. Secondly, the relationship between the music genre of the audio and its transparent bitrate is explored.

Most existing audio steganographic methods embed secret messages according to a pseudorandom number generator, thus some auditory sensitive parts in cover audio, such as mute or near-mute segments, will be contaminated, which would lead to poor perceptual quality and may introduce some detectable artifacts for steganalysis. In this paper, we propose a novel adaptive audio steganography in the time domain based on the advanced audio coding (AAC) and the Syndrome-Trellis coding (STC). The proposed method firstly compresses a given wave signal into AAC compressed file with a high bitrate, and then obtains a residual signal by comparing the signal before and after AAC compression. According to the quantity and sign of the residual signal, \(\pm 1\) embedding costs are assigned to the audio samples. Finally, the STC is used to create the stego audio. The extensive results evaluated on 10,000 music and 10,000 speech audio clips have shown that our method can significantly outperform the conventional \(\pm 1\) LSB based steganography in terms of security and audio quality.

MPEG-4 scalable lossless (SLS) coding is a unified solution for demands in high compression perceptual audio and high quality lossless audio. It provides a fine-grain scalable extension to the MPEG-4 advanced audio coding (AAC) perceptual audio coder up to fully lossless reconstruction. It is observed that the quality of SLS coded audio is still far from optimal when the core bitrate is low. With this observation, an adaptive bit-plane scanning (ABPS) method is proposed in this paper. In ABPS, the bit-plane scanning order is adaptive to the energy distribution of the signal to be coded. The results show that with ABPS, the perceptual quality of the audio under low core bitrate scenario is significantly improved without any extra payload.

Given the popularity of audio and video applications, compressed audio has become an important carrier of covert communication on the Internet. Many novel compressed audio steganography schemes have emerged that offer good hiding capability and aural concealment. In this paper, a universal steganalysis scheme called MultiSpecNet is proposed to detect steganography based on multiple embedding domains (advanced audio coding (AAC) and MPEG-1 Audio Layer III (MP3)), which are currently the two most popular compressed audio standards. The basic idea is that modification of either domain by a steganography scheme will change the time-frequency relationship of the audio signal after decoding. The proposed approach adopts the spectrogram as the input feature to extract richer information. DeepResNet is used to learn the distinguishing feature representations, and multiscale spectrograms are used to enrich the feature diversity. The experimental results show that the proposed scheme is effective at detecting different steganography schemes based on the AAC and MP3 embedding domains. The detection accuracy of the proposed scheme is higher than that achieved by other state-of-the-art schemes. Using spectrograms as the input, DeepResNet achieves better performance than schemes using quantized modified discrete cosine transform (MDCT) coefficients and mel-spectrogram, although the quantized MDCT coefficient is the parameter modified by the steganography schemes directly and mel-spectrogram is very popular and effective for general audio signal analysis. To the best of our knowledge, this work is the first audio steganalysis scheme that can detect multiple steganography schemes in both the MP3 and AAC embedding domains. The method proposed in this paper can be extended to audio steganalysis for other codecs or for audio forensics purposes.

Enhanced aacPlus is an audio codec which is composed of advanced audio coding (AAC), spectral band replication (SBR), and parametric stereo (PS) for efficient audio coding at low bit rates. We propose a new coding scheme for lossless bit rate reduction of PS in enhanced aacPlus. We first determine the optimal contexts for context-based coding of quantized stereo parameter indexes in PS. Then we propose a coding scheme which is based on context-based coding using the determined optimal contexts in conjunction with arithmetic coding. The proposed scheme has normal and memory-reduced versions, both of which are shown to guarantee significant and consistent lossless bit rate reduction of PS. The saved bits by the proposed scheme can be used for the other modules of enhanced aacPlus, AAC and SBR.

A magnetron is a power oscillator device used in several radar transmitters for generating the electromagnetic wave (EMW) with a constructive and fixed frequency. In this work, a radar transmitter has been carried out using magnetron operates at 2,458 MHz and two conducting layers waveguides to achieve minimum loss of the signal due to the increased temperature. The magnetron is connected with the high voltage capacitor (1.0 μF) in order to store the alternating current (AC) signal delivered from the high voltage transformer. A waveguide parts have been used as transmission lines in view of connecting the magnetron to the antenna. These parts include two metal layers, the upper layer (outer) is advanced audio coding (AAC) 1350 aluminum conductor and the lower (inner) is copper. The waveguide dimension should be suitable to flow frequency from 2.2 GHz to 3.3 GHz. To measure the operating frequency, a waveguide adapter and a coaxial cable which combined with n-female connector have been connected to magnetron. The wave is delivering to the frequency counter by Bayonet Neill–Concelman (BNC) to n-female connectors. The promising results of the proposed work have been achieved with a maximum power, efficient return loss, acceptable voltage standing wave ratio (VSWR) and low-cost manufacturing.

The tones of existing instruments are simulated in the time domain by means of standard finite difference methods. The string model previously applied to the guitar [A. Chaigne, J. Acoust. Soc. Am. Suppl. 1 88, S188 (1990)] and more recently to the piano [A. Chaigne, A. Askenfelt, and E. Jansson, J. Acoust. Soc. Am. 89, 1878(A) (1991)] has been extended to the violin, the xylophone bars, and tympani membranes. The general procedure, which is similar for each type of instrument, will be presented. It consists first in deriving a numerical scheme from the continuous differential equations governing the motion of the vibrating systems including the interaction with an exciter (hammer, bow, mallets,...). In a second step, the mathematical properties of these discrete approximations are evaluated with special emphasis laid on stability, accuracy, and dispersion problems. Simulated waveforms are computed with realistic values of the parameters derived from measurements on existing instruments. These waveforms as well as their spectral contents are then systematically compared with those of real tones. This time-domain approach is used in order to test the influence of various physical parameters on the transients of strings and percussive sounds in which the nonlinear effects are essential. The paper will be illustrated by sound synthesis examples of various instruments obtained with this technique.

Bandwidth efficiency and error robustness are two essential and conflicting requirements for streaming media content over error-prone channels, such as wireless channels. This paper describes a new scheme called content-based unequal error protection (C-UEP), which aims to improve the user-perceived QoS in the case of packet loss. We use music streaming as an example to show the effectiveness of the new concept. C-UEP requires only a small fraction of the redundancy used in existing forward error correction (FEC) methods. C-UEP classifies every audio segment (e.g. an encoding frame) into different classes to improve encoding efficiency. Salient transients such as drumbeats and note onsets are encoded with more redundancy in a secondary bitstream used to recover lost packets by the receiver. Formal perceptual evaluations show that our scheme improves audio quality significantly over simple muting and packet repetition baselines. This improvement is achieved with a negligible amount of redundancy, which is transmitted to the receiver ahead of playback.

Omnidirectional video (ODV) streaming applications are becoming increasingly popular. They enable a highly immersive experience as the user can freely choose her/his field of view within the 360-degree environment. Current deployments are fairly simple but viewport-agnostic which inevitably results in high storage/bandwidth requirements and low Quality of Experience (QoE). A promising solution is referred to as tile-based streaming which allows to have higher quality within the user’s viewport while quality outside the user’s viewport could be lower. However, empirical QoE assessment studies in this domain are still rare. Thus, this paper investigates the impact of different tile-based streaming approaches and configurations on the QoE of ODV. We present the results of a lab-based subjective evaluation in which participants evaluated 8K omnidirectional video QoE as influenced by different (i) tile-based streaming approaches (full vs. partial delivery), (ii) content types (static vs. moving camera), and (iii) tile encoding quality levels determined by different quantization parameters. Our experimental setup is characterized by high reproducibility since relevant media delivery aspects (including the user’s head movements and dynamic tile quality adaptation) are already rendered into the respective processed video sequences. Additionally, we performed a complementary objective evaluation of the different test sequences focusing on bandwidth efficiency and objective quality metrics. The results are presented in this paper and discussed in detail which confirm that tile-based streaming of ODV improves visual quality while reducing bandwidth requirements.