An encoding device (200) includes an MDCT unit (202) that transforms an input signal in a time domain into a frequency spectrum including a lower frequency spectrum, a BWE encoding unit (204) that generates extension data which specifies a higher frequency spectrum at a higher frequency than the lower frequency spectrum, and an encoded data stream generating unit (205) that encodes to output the lower frequency spectrum obtained by the MDCT unit (202) and the extension data obtained by the BWE encoding unit (204). The BWE encoding unit (204) generates as the extension data (i) a first parameter which specifies a lower subband which is to be copied as the higher frequency spectrum from among a plurality of the lower subbands which form the lower frequency spectrum obtained by the MDCT unit (202) and (ii) a second parameter which specifies a gain of the lower subband after being copied.

Encoding device and decoding device

A system, computer-readable medium and generated signal are disclosed for extending the bandwidth of a first signal (i.e., a narrowband signal) such as a speech signal. The system produces a second signal from a first signal by computing first area coefficients from a first signal, generating second area coefficients from the first area coefficients and generating a second signal using the second area coefficients. The first signal may be a narrowband signal and second signal may be a wideband signal. The first area coefficients may be narrowband coefficients and the second area coefficients may be wideband area coefficients.

System for bandwidth extension of narrow-band speech

A system and method are disclosed for extending the bandwidth of a narrowband signal such as a speech signal. The method applies a parametric approach to bandwidth extension but does not require training. The parametric representation relates to a discrete acoustic tube model (DATM). The method comprises computing narrowband linear predictive coefficients (LPCs) from a received narrowband speech signal, computing narrowband partial correlation coefficients (parcors) using recursion, computing Mnb area coefficients from the partial correlation coefficient, and extracting Mwb area coefficients using interpolation. Wideband parcors are computed from the Mwb area coefficients and wideband LPCs are computed from the wideband parcors. The method further comprises synthesizing a wideband signal using the wideband LPCs and a wideband excitation signal, highpass filtering the synthesized wideband signal to produce a highband signal, and combining the highband signal with the original narrowband signal to generate a wideband signal. In a preferred variation of the invention, the Mnb area coefficients are converted to log-area coefficients for the purpose of extracting, through shifted-interpolation, Mwb log-area coefficients. The Mwb log-area coefficients are then converted to Mwb area coefficients before generating the wideband parcors.

A method of bandwidth extension for narrow-band speech

Recommendation G.729.1 is a new ITU-T standard which was approved in May 2006. This recommendation describes a hierarchical speech and audio coding algorithm built on top of a narrowband core codec. One challenge in the codec design is the generation of a wideband signal with a very limited additional bit rate (less than 2 kb/s). In this paper, we describe the respective codec layer, which extends the transmitted acoustic bandwidth from the narrowband frequency range (50 Hz-4 kHz) to the wideband frequency range (50 Hz-7 kHz). The underlying algorithm uses a fairly coarse parametric description of the temporal and spectral energy envelopes of the high frequency band (4-7 kHz). This parameter set is quantized with a bit rate of 1.65 kb/s. At the decoder side, the high-frequency components are regenerated by appropriately shaping a synthetically generated ldquoexcitation signal.rdquo Apart from the algorithmic description and a discussion, we state a complexity evaluation as well as some listening test results.

Bandwidth Extension for Hierarchical Speech and Audio Coding in ITU-T Rec. G.729.1

The limited audio bandwidth used in narrowband telephone systems degrades both the quality and the intelligibility of speech. This paper presents a new method for the bandwidth extension of telephone speech. Frequency components are added to the frequency band 4-8 kHz using only the information in the narrowband speech. A neural network is used to estimate the mel spectrum in the extension band in short time frames based on features calculated from the narrowband speech. A wideband excitation signal is generated by spectral folding from the narrowband linear prediction residual and a filter bank is utilized to divide the excitation into four sub-bands that cover the extension band. These sub-bands are weighted such that the estimated mel spectrum is realized. Bandwidth-extended speech is obtained by summing the weighted sub-bands and the original narrowband signal. Listening tests show that this new method improves speech quality compared with narrowband telephone speech and with a previously published bandwidth extension method.

Bandwidth Extension of Telephone Speech Using a Neural Network and a Filter Bank Implementation for Highband Mel Spectrum

In this paper, an encoding technique called Hi-BIN (High Band Injection), which can be combined with any narrowband coder to achieve good quality wideband speech, is described. The principle behind this technique is to model frequencies above 4 kHz by noise with an appropriate spectral shape. This simple way of injecting synthetic noise in the higher frequencies gives surprisingly good quality when compared to very widely used computationally intensive waveform coding techniques such as CELP. We show that Hi-BIN offers a low bit-rate representation of the higher band and is backwards compatible with existing narrowband speech coding systems.

Hi-BIN: an alternative approach to wideband speech coding

We propose methods to achieve bit-rate and complexity scalability in CELP coders/decoders. These methods can be implemented as plug-ins or add-ons to existing CELP systems. The associated scale factors can be readjusted even during run-time. In particular, using the proposed methods, a wideband (sampling rate 16 kHz) coder/decoder has been developed that is capable of reconstructing good to high quality speech at bit rates ranging from 14 to 24 kbit/s. The possible complexity range of the decoder can amount to a factor of 2 or more, depending on the constitution of the transmitted bitstream. Although the system also incorporates a complexity scalable encoder, it is not discussed in this paper. This system has been submitted to MPEG-4, where it has been incorporated in the MPEG-4 Audio Verification Model.

R. Taori

Papers

Hi-BIN: an alternative approach to wideband speech coding

On scalability in CELP coding systems