High Quality Speech Coding Research Articles

English Listening Teaching Mode under Artificial Intelligence Speech Synthesis Technology

To discuss the application of artificial intelligence (AI) speech synthesis technology and wireless network technology in English listening teaching, firstly, the present situation of English listening teaching is analyzed. Secondly, a questionnaire is designed to investigate the feasibility of AI speech synthesis technology in English listening teaching. Finally, the control group (a traditional mode of English listening teaching) and experimental group (AI speech synthesis technology) are set up, compared, and analyzed. The survey shows that for natural fluency, speech synthesis has the lowest score. The textbook audio score is the highest, with an average score of 4.23. According to the mean opinion score (MOS), the AI speech synthesis technology belongs to high-quality speech coding. Therefore, the English audio generated by combining AI speech synthesis technology under wireless network technology also has high quality. Meanwhile, 113 people in the survey consider that the clear articulation of the synthesized audio is full marks, which is clearer than the English audio matching the textbook, indicating that students think the English audio of the supporting textbook is poor in the clear articulation, which affects students’ listening. Moreover, the English listening scores of the experimental group are notably better than those of the control group. It is hoped that the traditional teaching mode can be better integrated with wireless networks and AI technology, thereby improving the clarity of English listening, providing mobile learning, and advancing students' English performance and learning efficiency.

ESTIMATION OF PULSED SPEECH MODEL PARAMETERS

Methods for estimating speech model parameters are disclosed. For pulsed parameter estimation, a speech signal is divided into multiple frequency bands or channels using bandpass filters. Channel processing reduces sensitivity to pole magnitudes and frequencies and reduces impulse response time duration to improve pulse location and strength estimation performance. These methods are useful for high quality speech coding and reproduction at various bit rates for applications such as satellite and cellular voice communication.

Adaptive nonuniform polar quantization application to high quality speech compression

The logarithmic companding technique has shown to be extremely useful in speech quantization with rate of 8 bits/sample. However, for lower bit rates it is not the ideal solution for high quality speech coding. Because of that, in this paper we establish source coding scheme which enables better spectrum efficiency for input that has a large dynamic range. Since our wish is also to improve signal quality in comparison with quality defined with standards G.711 and G.712, we opt for adaptive technique application to the speech coding. Our research shows that proper design of forward gain-adaptive polar quantization can enable compression of about 1 bit/sample as well as significantly better quality than in case of using coder designed according to standard G.711. Furthermore, performances can be sustained over the whole speech dynamic range. Also, if the requisite speech quality is not supposed to be lower than G.712 standard quality, the achieved compression can be almost 1.5 bits/sample. Besides, we propose knew simple encoding rule which can additionally reduce bit rate for 0.1 bit/sample.

Multi-segment vector quantizer for a speech coder suitable for use in a radiotelephone

A Vector-Sum Excited Linear Predictive Coding (VSELP) speech coder provides improved quality and reduced complexity over a typical speech coder. VSELP uses a codebook which has a predefined structure such that the computations required for the codebook search process can be significantly reduced. This VSELP speech coder uses single or multi-segment vector quantizer of the reflection coefficients based on a Fixed-Point-Lattice-Technique (FLAT). Additionally, this speech coder uses a pre-quantizer to reduce the vector codebook search complexity and a high-resolution scalar quantizer to reduce the amount of memory needed to store the reflection coefficient vector codebooks. Resulting in a high quality speech coder with reduced computations and storage requirements.

Time-frequency interpolation with application to low rate speech coding

A new method for high quality speech coding, Timing-Frequency Interpolation (TFI) which offers advantages over conventional CELP (code-excited linear predictive) algorithms for low rate coding. The method, provides a perceptually advantageous framework for voiced speech processing. The general formulation of the TFI technique is described.

Adaptive forward-backward quantizer for low bit rate high-quality speech coding

A novel variable-rate linear predictive coding (LPC) parameter quantization scheme is proposed, in which linear prediction is done by using either the current (forward LPC) or previously decoded (backward LPC) speech blocks. The proposed LPC quantization scheme was integrated into the FS1016 Federal Standard CELP coder. Significant LPC bit rate reduction is achieved without compromising the decoded speech quality.

Current objectives in 4-kb/s wireline-quality speech coding standardization

The standardization of high-quality speech coding has intensified. In parallel, a number of novel applications are placing new demands on transmission efficiency and quality. In response to such challenges, standardization bodies have begun the definition of requirements for the next generation of very low-rate speech coding. Taking a lead in these activities, ANSI committee T1A1 and the ITU-T initiated the definition of the performance and characteristics of a wireline-quality 4-kb/s speech coding algorithm for network applications. This emerging set of requirements is presented.< <ETX xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">&gt;</ETX>

On encoding pitch and LPC parameters for low‐rate speech coders

AbstractThis paper describes the development of efficient methods to encode the parameters of long‐term (pitch) and short‐term (LPC all‐pole) prediction filters with a limited bit‐rate, such that the design of high quality speech coders in the 3–4 kbit/s range becomes easier. Our results indicate that good performance can be reached with about 800 bit/s to represent both the pitch lags and the coefficients of a three‐tap pitch filter, thus achieving a significant 36% bit‐rate reduction in comparison with conventional quantization techniques, and using 24 bit/frame to encode spectral parameters relative to a multilanguage speech database.

Performance of adaptive prediction algorithms for trellis coded quantization of speech

Investigates the performance of different adaptive prediction algorithms for use in a low delay, high-quality speech coder at 18 kbit/s. The system to be studied is based on trellis coded quantization (TCQ). In the work presented, the authors demonstrate that when trellises based on feedback-free convolutional encoding circuits are employed, error propagation in TCQ systems is not a serious problem. Performance comparisons are made using a fixed predictor and four different adaptive predictors. Informal listening tests reveal that the encoded speech is of good communications quality at an error rate of 10/sup -3/. In an ideal channel, a system employing an all-pole least squares lattice predictor with pitch prediction produces reconstructed speech that is nearly indistinguishable from the original.< <ETX xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">&gt;</ETX>

Shared time division duplexing: an approach to low-delay high-quality wireless digital speech communications

Various strategies to provide low-delay high-quality digital speech communications in a high-capacity wireless network are examined. Various multiple access schemes based on time-division and packet reservation are compared in terms of their statistical multiplexing capabilities, sensitivity to speech packet dropping, delay, robustness to lossy packet environments, and overhead efficiency. In particular, a low-delay multiple access scheme, called shared time-division duplexing (STDD) is proposed. This scheme allows both the uplink and downlink traffic to share a common channel, thereby achieving high statistical multiplexing gain even with a low population of simultaneous conversations. The authors also propose a choice of low delay, high quality speech coding and digital modulation systems based on adaptive DPCM, with QDPSK or pseudo-analog transmission (skewed DPSK), for use in conjunction with the STDD multiple access protocol. The choice of the alternative systems depends on required end-to-end delay, recovered speech quality and bandwidth efficiency. Typically, with a total capacity of 1 MBaud, 2 ms frame and 8 kBaud speech coding rate, low delay STDD is able to support 48 pairs of users compared to 38, 35, and 16 for TDMA with speech activity detection, basic TDMA and PRMA respectively. This corresponds to respective gains of 26%, 37% and 200%.< <ETX xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">&gt;</ETX>