Adaptive Multi-Rate audio codec

About: Adaptive Multi-Rate audio codec is a research topic. Over the lifetime, 1467 publications have been published within this topic receiving 19736 citations. The topic is also known as: AMR & Adaptive Multi-Rate.

Golomb-rice coding optimized via LPC for frequency domain audio coder

Abstract: We have devised a method to optimize Golomb-Rice coding of frequency spectra, aiming at its use in frequency domain audio coder, using spectral envelopes extracted by linear predictive coding (LPC) from amplitude spectra instead of conventional power spectra according to theoretical investigations This optimization improves the efficiency of the Golomb-Rice coding by allocating Rice parameter at each frequency bin based on the value of the envelopes, resulting in the enhancement of the objective and subjective quality of the state-of-the-art wideband coder at 16 kbit/s Therefore, the method introduced here is expected to be useful for coding audio signals at low-bit-rate and low-delay conditions, required in mobile communications

Fragile speech watermarking scheme with recovering speech contents

Abstract: The wide use of digital speech recorders becomes a serious matter when they are involved in assisting with court rulings. How to distinguish if a recorded content is valid or not becomes a life-or-death question. In light of this concern, least significant bits (LSB) of excitation signals would be used as fragile watermarks in the hybrid speech vocoder. In addition, a location-variable content-dependent watermark generating mechanism is proposed. Such location-variable content-based watermark would allow users to detect where in the recording the content is being replaced, inserted, or deleted. Lastly, an attempt is done to store partial reconstruction data in the LSBs of excitation signals in the G.723.1 speech codec, so that the original speech content may be reconstructed after counterfeited. The proposed system is demonstrated to be a reliable system, with test results showing that a recording with watermarks has a perceptual evaluation of speech quality (PESQ) value down 0.2, while the accuracy in detecting faked regions can be up to 97.45%.

Network functions for the application of multi-rate speech coding in digital cellular systems

Abstract: Multi-rate speech coding is a considerable subject in digital cellular systems using highly efficient speech coding methods. The paper describes network functions for the application of the multi-rate speech coding network. As an example of multi-rate coding, functions of a half rate speech coding method in personal digital cellular (PDC) systems are mainly discussed. One of the functions is a codec bypassed connection control to reduce the degradation of voice quality in communication from a mobile station to a mobile station. We propose speech coding negotiation schemes between the originating and terminating MS for the codec bypassed connection control. We also propose a non-voice service connection control scheme which is necessary to provide non-voice services with the same quality as a full rate coding. >

Backward adaptive warped lattice for wideband stereo coding

Abstract: In this paper an extremely low delay perceptual audio codec is presented. The codec is based on warped linear prediction which inherently utilizes auditory frequency resolution and frequency masking characteristics of hearing. In the current version of the codec the coding delay is the minimum. This is achieved using backward adaptive lattice methods where waveform modeling is completely based on already transmitted data. Coding technique is applied separately to the two channels but the quantization processes are unified to gain more bit rate reduction.

Codec deployment using in-band signals

Abstract: After a call is established between two stations using a codec that has been negotiated during call setup, in-band signaling may be used between the two stations to change the codec that is to be used. The in-band signals are indicative that the station that is transmitting the in-band signals can operate with a second codec and are used to probe whether the receiving station can also operate with that second codec. If the receiving station detects and reacts to the in-band signals, then both stations change to communicate with the second codec. The second codec has compatible packet sizes of the deployed (originally negotiated) codec without any need of infrastructure upgrade and/or quality compromise to legacy phone users (i.e., stations that cannot operate with the second codec).