## 4 Applications in Multi-Channel Speech Coding and Enhancement
### Application in Multi-Channel Speech Coding  
The enhance voice services (EVS) [23] is a new kind of speech coding standard after adaptive multi-rate (AMR) [24] and adaptive multi-rate wideband (AMR-WB) [25]. It has gradually replaced AMR and AMR-WB as a mainstream trend on mobile devices such as mobile phones. Compared with existing coding standards, the EVS codec enhanced quality and coding efficiency for narrowband (NB) and wideband (WB) speech services and enhanced quality by the introduction of super wideband (SWB) and full band (FB). Furthermore, the EVS codec has backward compatibility to the AMR-WB codec [26]. Given the EVS codec, a multi-channel speech codec based on the EVS and time delay of the TDOA estimation is proposed. That is, the speech signal of reference channel is encoded by the EVS codec, and the time delay between other channels and reference channel is estimated and used to represent spatial information. The encoded speech bitstream and the coded time delay bitstream are integrated to form final transmission bitstream. At the decoder, the speech bitstream and the time delay bitstream are separated and decoded, and finally, the multi-channel output speech is recovered. In order to get a more accurate time delay, the linear forward spatial prediction-based TDOA estimation method [27] is used. The block diagram of the proposed multi-channel speech coding method combining with EVS and TDOA (called Multi-EVS-TDOA) are shown in Figure 10.  
Once the time delay is estimated, it is quantized by uniform quantization method. As shown in Figure 3, the speech source can appear at any position relative to the microphone array. Under the far-field assumption, the time delay between the microphones is 0 when the speech source is facing the microphone array. When the speech source is in line with the microphone array, the time delay between the microphones reaches the maximum. So, for the uniform linear array in this work, the distance between microphones is 0.04 m, and the maximum value of time delay is 2 samples. In this way, a 2-bit uniform quantizer with four levels can be used to quantize the estimated time delay that is the spatial information of speech source. Thus, its bit rate is calculated as (2560 bits/frame \(\times\) 1 channel + 2 bits/frame) \(\times\) 50 frames/s = 128.1 kbps, where 2560 is the bit number of one frame of EVS codec at 128 kbits/s, 2 is the bit number of one frame for the time delay, and 50 is the frame rate. The performance bottleneck of the Multi-EVS-TDOA method is the accuracy of time delay estimation. In the current room acoustic environment, as shown in Table 2, the time delay should be 1.3113 samples. Because the time delay can only be estimated as an integer value, we have to manifold up-sample signal such as 10 times and estimate time delay. This up-sampling method improves the accuracy of time delay estimation, reduces the error, and increases speech quality. The block \begin{table}
\begin{tabular}{c c c} \hline \hline
**Application** & **Multi-Channel Speech Coding** & **Multi-Channel Speech Enhancement** \\ \hline \multirow{2}{*}{Input} & \multicolumn{2}{c}{Noisy speech} \\  & Clean speech with reverberation \\  & & Noisy speech with reverberation \\ \hline \hline \end{tabular}
\end{table} 
Table 4: Applications of the MASS. Figure 10: Block diagram of the Multi-enhance voice services (EVS)-TDOA. diagram of the proposed up-sampling-based multi-channel speech coding method combining with EVS and TDOA (called US-Multi-EVS-TDOA) are shown in Figure 11. For the proposed multi-channel speech coding method, the sampling rate can be seen as a priori knowledge, and for multi-channel speech with a specific sampling frequency, the proposed multi-channel speech coding method based on the up-sampling can be directly used for the encoding and decoding. Due to the up-sampling, the maximum value of delay goes up to 20. These result in more bit requirements for the quantization of time delay. Here, a 5-bit uniform quantizer with 32 levels is used for quantizing time delay.