In order to effectively utilize radio wave resources in a mobile communication system, it is required to compress speech signals at a low bit rate. On the other hand, it is expected from the user to improve quality of communication speech and implement communication services with high presence. In order to implement this, it is preferable not only to improve quality of speech signals, but also to be capable of encoding signals other than speech, such as audio signals having a wider band with high quality.
Further, in an environment where various types of networks are present, a speech coding scheme is required that can flexibly support communication between different networks, communication between terminals utilizing different services, communication between terminals having different processing performance, and conversational communication at multipoints as well as communication between two parties.
Moreover, a speech coding scheme is required to be robust against transmission path errors (in particular, packet loss in packet switching networks typified by IP networks).
One speech coding scheme satisfying such requirements is the bandwidth scalable speech coding scheme. The bandwidth scalable coding scheme is a coding scheme that encodes speech signals in a layered way, and a coding scheme where coding quality increases in accordance with an increase in the number of coding layers. The bit rate can be set variable by increasing or decreasing the number of coding layers, so that it is possible to effectively use transmission path capacity.
Further, with the bandwidth scalable speech coding scheme, it is only necessary to receive at least the data coded by a base layer at a decoder side, and it is possible to allow to some extent information coded by additional layers being lost on the transmission path, and therefore the bandwidth scalable speech coding scheme provides robustness against transmission path errors. Further, the frequency bandwidth of speech signals to be encoded also becomes wider in accordance with an increase in the number of coding layers. For example, for a base layer (i.e. core layer), a coding scheme for telephone band speech of the related art is used. Further, in additional layers (i.e. enhancement layers), layers are configured so that wideband speech which has a bandwidth such as 7 kHz can be encoded.
In this way, with the band scalable speech coding scheme, telephone band speech signals are encoded in the core layer, and high-quality wideband signals are encoded in the enhancement layers, so that it is possible to utilize the bandwidth scalable speech coding scheme for both telephone band speech service terminals and high-quality wideband speech service terminals and support multipoint communication including the two kinds of terminals. Further, the coded information is layered, so that it is possible to increase error robustness by devising a transmission method, and readily control the bit rate on the encoding side or on the transmission path. Therefore, the bandwidth scalable speech coding scheme draws attention as a speech coding scheme for future communication.
The method disclosed in non-patent document 1 is given as an example of the bandwidth scalable speech coding scheme described above.
In the bandwidth scalable speech coding scheme disclosed in non-patent document 1, MDCT coefficients are encoded using a scale factor and fine structure information for each band. The scale factor is Huffman encoded, and the fine structure is subjected to vector quantization. Auditory weighting of each band is calculated using a scale factor decoding result, and the bit allocation to each band is decided. The bandwidth of each band is non-uniform and set in advance so as to become wider for a higher band.
Further, transmission information is classified into four groups as described below.
A: Core codec coding information
B: High-band scale factor coding information
C: Low-band scale factor coding information
D: Spectrum fine structure coding information
Further, the following processing is carried out on the decoding side.
<Case 1> When information for A cannot be received completely, decoded speech is generated by carrying out frame erasure concealment processing.
<Case 2> When only information for A is received, a decoded signal for the core codec is outputted.
<Case 3> When information for B is received in addition to the information for A, a high band is generated by mirroring the decoded signal for the core codec and a decoded signal having a wider bandwidth than the decoded signal of the core codec is generated. Decoded information for B is used in generation of high band spectrum shapes. Mirroring is carried out at a voiced frame, and is carried out so that the harmonic structure does not collapse. The high band is generated at an unvoiced frame using random noise.<Case 4> When information for C is received in addition to information for A and B, the same decoding processing as in case 3 is carried out using only information for A and B.<Case 5> When information for D is received in addition to the information for A, B and C, complete decoding processing is carried out at bands where all information for A to D is received, and a fine spectrum is decoded by mirroring a decoded signal spectrum on the low band side at bands where information for D is not received. Even if the information for D is not received, it is possible to receive the information for B and C, and this information for B and C is utilized in decoding of spectrum envelope information. Mirroring is carried out at a voiced frame, and is carried out so that the harmonic structure does not collapse. The high band is generated at an unvoiced frame using random noise.Non-patent document 1: B. Kovesi et al, “A scalable speech and audio coding scheme with continuous bit rate flexibility,” in proc. IEEE ICASSP2004, pp. I-273--I-276.