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Showing papers on "Enhanced Variable Rate Codec published in 2017"


Journal ArticleDOI
TL;DR: There are no significant benefits to lowering the mode-sets or deploying dynamic codec rate adaptation, and controlled laboratory experiments indicated that there was an improvement in voice quality when mode-set eight was employed.
Abstract: In this paper, we examine the impact of four voice over long term evolution adaptive multi-rate wideband codec mode-sets on coverage at pedestrian and vehicular speeds Industry-standardized mean opinion scores were used as a metric for voice quality Controlled laboratory experiments simulating pedestrian speeds indicated that there was an improvement in voice quality when mode-set eight was employed At vehicular speeds, mode-set eight outperformed the other mode-sets for path losses less than 130 dB; however, all four mode-sets experienced a significant decline in voice quality when the path loss was greater than 130 dB Based on the current implementations, there are no significant benefits to lowering the mode-sets or deploying dynamic codec rate adaptation

13 citations


Journal ArticleDOI
TL;DR: The proposed codec CPCM (Chaotic Pulse Code Modulaion) will join the encryption to the compression of the voice data, which provides the same compression ratio given by the PCM codec, but with an unintelligible content.
Abstract: We propose to incorporate encryption procedure into the lossy compression of voice data PCM(Pulse Code Modulation) based on the A-law approximation quantization. The proposed codec CPCM (Chaotic Pulse Code Modulaion) will join the encryption to the compression of the voice data. This scheme provides the same compression ratio given by the PCM codec, but with an unintelligible content. Comparisons with many used schemes have been made to highlight the proposed method in terms of security and rapidity. CPCM codec can be a better alternative to Compress-then-encrypt classical methods which is a time and resource consuming and non suitable for real-time multimedia secure transmission.

12 citations


Patent
19 Oct 2017
TL;DR: In this paper, the authors proposed a method for enhanced codec control in wireless networks, where the UE determines all available codec bitrates that can be performed by the UE and communicates the bitrates before receiving the codec control command.
Abstract: Apparatus and methods are provided for enhanced codec control. In one novel aspect, a method includes receiving a codec control command by a user equipment (UE) in a wireless network, determining if the recommended codec characteristic will be applied to a codec executing on the UE, and adjusting a characteristic of the codec executing on the UE based on the recommended codec characteristic. The UE is connected with a radio access network (RAN) and the codec control command includes a recommended codec characteristic. In another novel aspect, the recommended codec characteristic is a maximum bitrate. In another embodiment, the recommended codec characteristic is a type of codec. In yet another novel aspect, the recommended codec characteristic is a radio resource allocation command. In another novel aspect, the UE determines all available codec bitrates that can be performed by the UE and communicates the bitrates before receiving the codec control command.

6 citations


Journal ArticleDOI
TL;DR: The testing results show that in the same bit-rate, the quality of the proposed BWE is better than the BWE method in Audio Video Standard (AVS) Part 10, and the computational complexity reduced evidently.
Abstract: To meet the requirements of low coding bit-rate and low complexity for audio coding in mobile surveillance device, in this paper we proposed an audio Bandwidth Extension (BWE) algorithm based on a hybrid prediction model including the intra-frame prediction, inter-frame prediction and white-noise prediction. In the algorithm, we used four different predicting modes, including translation and fold of low-frequency signals, high-frequency signal in previous frame, and white noise, to reconstruct high-frequency signals for various types of audio signals. When a high frequency frame is encoded, Signal-Noise-Ratios (SNR) of all modes are calculated and compared with each other. The prediction pattern with highest SNR is selected as the most accurate prediction pattern to encoding the high frequency signal. And two indicator bits are used indicate the encoding mode. When the compressed high frequency frame is decoded, the specific mode is selected based on the indicator bits. The testing results show that in the same bit-rate, the quality of the proposed BWE is better than the BWE method in Audio Video Standard (AVS) Part 10, and the computational complexity reduced evidently. These advantages help the proposed method to meet the requirements of mobile surveillance device for audio codec.

3 citations


Book ChapterDOI
01 Jan 2017
TL;DR: A new Analytic Hierarchy Process (AHP)-based CAC algorithm is proposed for increasing the number of admission and reducing the admission of less compressed calls to give better QoS performance for real-time services and increase the system throughput for Voice over Internet Protocol.
Abstract: Call Admission Control (CAC) in wireless communication plays major role in deciding the admission of real-time and non real-time mobile users. For non real-time, it does not care for the Quality of Service (QoS) performance. But for real-time services, the CAC cares for the QoS by keeping the sufficient bandwidth throughout the transmission. Bandwidth determines the system capacity and speed. In this paper, a new Analytic Hierarchy Process (AHP)-based CAC algorithm is proposed for increasing the number of admission and reducing the admission of less compressed calls. For this task the various codecs such as G.711, G.729, G.723, G.726, AMR, EVRC, and iLBC have been taken. AHP is applied among them to take the right decision and it produces the ranking order. That rank helps to save more bandwidth and increases the throughput (Packets per second). This task is carried out by taking the criteria like bandwidth, packetization delay, and compression ratio of each individual codecs. The ultimate aim of this TACA is to give better QoS performance for real-time services and increase the system throughput for Voice over Internet Protocol.

2 citations


Proceedings ArticleDOI
20 Apr 2017
TL;DR: The article describes the results of research on the selection of these bits of the codebook codec G.729 which the negation of the least have influence to the loss of quality and fidelity of the output signal.
Abstract: Network steganography is dedicated in particular for those communication services for which there are no bridges or nodes carrying out unintentional attacks on steganographic sequence. In order to set up a hidden communication channel the method of data encoding and decoding was implemented using code books of codec G.729. G.729 codec includes, in its construction, linear prediction vocoder CS-ACELP (Conjugate Structure Algebraic Code Excited Linear Prediction), and by modifying the binary content of the codebook, it is easy to change a binary output stream. The article describes the results of research on the selection of these bits of the codebook codec G.729 which the negation of the least have influence to the loss of quality and fidelity of the output signal. The study was performed with the use of subjective and objective listening tests.