Clear Water Bay, Kowloon, HONG KONG. ABSTRACT. A new unequal error protection scheme for enhancing the quality of the wireless transmitted MPEG audio ...
UAL ERROR PROTECTION FOR WIRELESS TRANSMISSION OF MPEG AUDIO* Chi Wai Yung , Hung Fai Fu, Chi Ying Tsui, Roger S. Cheng, Doug George Consumer Media Laboratory Department of Electrical & Electronic Engineering The Hong Kong University of Science & Technology Clear Water Bay, Kowloon, HONG KONG
ABSTRACT A new unequal error protection scheme for enhancing the quality of the wireless transmitted MPEG audio is introduced. The scheme is divided into two parts. The first part, namely the unequal frame protection, put more protection on the header portion of the MPEG frame and less protection on the data sample portion of the MPEG frame. The second scheme, namely the unequal sample protection, put more protection on the most significant bits '(MSBs) of the quantized data samples and less protection on the least significant bits (LSBs). Simulation results show that when comparing to the equal error protection scheme, the proposed scheme reduces the frame error rate by a factor of 2 and lowers the peak-subband-error-power-to-mask ratio of the audio data by as much as 11 dB.
1. INTRODUCTION Technological advancement in wireless communications in recent years enables the wireless transmission of not only low quality speech, but also high fidelity multimedia data. Among various applications, the more popular ones include Digital Audio Broadcasting (DAB) and Digital Wireless Audio (DWA) environment at home, which deliver high quality audio over wireless environment in real-time. The most important performance measures in wireless communication system design are the bit error rate (BER) and the frame error rate (FER). While these are meaningful measures for most computer applications such as file transfer, they do not necessarily reflect the perceptual quality of multimedia data. In particular, for the highly compressed audio or video signals, a very rare (by wireless system standard) bit error can lead to a highly annoying or very unacceptable perceptual distortion. Instead of demanding a very low BER which is expensive to achieve in wireless environment, the use of unequal error protection or joint design of source and channel coders have been studied [ l ] and shown to be very promising in achieving good perceptual quality without exceedingly complex signal processing. In this paper, we propose a new unequal error protection scheme for a wireless digital audio transmission system that delivers high quality audio signal within an indoor environment over t h e unlicensed ISM bands. The major impairments in this type of wireless channels are fading and potential narrowband interference generated by other products using the same band. To lessen the effect of deep fade and interference, Reed Solomon code [2] is used with a fast frequency hopping scheme. An interleaver is designed to distribute the symbols of a RS
*This work is supported in part by Hong Kong Industrial Support Fund under project number AF264197.
codeword over multiple hops, allowing the error correcting capability of the RS code to correct symbols corrupted by interference and deep fades. The hopping has to be fast to reduce the end-to-end delay and the size of the interleaver. We assume that the audio signal is compressed using the MPEG I layer 2 audio standard [3,4,5].In that scheme, each compressed frame consists of two parts: the header and the data samples. Even with the Reed Solomon code, error can still be found in the received audio data. In order to achieve high fidelity audio data, we propose a new unequal error protection coding scheme to improve the quality of the received audio data. The scheme is divided into two parts: the unequal frame protection and the unequal sample protection. The main idea of the unequal frame protection is to use a strong code with more redundancy bits to protect the header portion of the MPEG frame and few redundancy bits for the data sample portion of the MPEG frame. The main idea of the unequal sample protection is to use more redundancy bits to protect the MSB and less redundancy bits to protect the LSB of each data sample.
2. BACKGROUND MPEG/Audio is an example of subband coding [3]. Although it is a lossy compression algorithm, it give a transparent and perceptually lossless compression. The encoder is divided into 4 parts, a filter bank, a psychoacoustic model, a bit allocation/quantizer/coding block, and a bitstream formatting block. When the encoder receives the audio signal in pulse code modulation (PCM) format, it passes the signal to the filter bank which divides the audio signal into 32 equal-width frequency subbands. For each audio channel, a block of N temporal samples is then converted into N new samples spread across the audio subbands with N/32 samples per subband. At the same time, the encoder passes the audio signals to the psychoacoustic model. The psychoacoustic model takes into account the characteristics of the human auditory system's inability to hear quantization noise under conditions of auditory masking. Whenever there is a strong audio signal, weaker audio signals at the temporal or spectral neighborhood will be masked off and become imperceptible. Using this property, the model analyzes the audio signal, determines the masking thresholds using psychoacoustic information and allocates bits to each subband such that each of the subband samples is quantized and encoded so as to keep the quantization noise below the masking threshold. The quantized audio signal data are then assembled into frames. The frame is divided into two parts: the header and the sample. The header includes information such as CRC, bit allocation information, scale factor selection information and scale factor. The sample portion includes the de-normalized data from each subband. The MPEG/Audio decoder is divided into three parts, a bitstream unpacking block, a frequency sample reconstruction block and a
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frequency-to-time mapping block. When the decoder receives the MPEG/Audio bitstream, the MPEG frames are unpacked. Using the unpacked data, i.e. bit allocation information, SCSFI, scale factors and data samples, the decoder reconstructs tlhe audio signal back to 32 equal-width frequency subbands. Lastly, a frequency-to-time mapping turns the subband signals back into the PCM format.
3. DIGITAL WIRELESS AUDIO SYSTEM There are four blocks in the encoder of the system, i.e. MPEG/Audio encoder, source-based interleaver, F!S code encoder and block interleaver. At the decoder side, there are block deinterleaver, RS code decoder, source-based deinterleaver and MPEG/Audio decoder. The connection between the encoder and the decoder is a wireless channel model based on a fast frequency hopping system over the ISM band. Hence, at each hop, the channel may suffer from interference with a probability equal to the ratio of the bandwidth with interference to the total bandwidth in the ISM band. The MPEG/Audio encodeddecoder is used for audio source coding. The bit rate of the encoder MPEG/Audio frame is 128kbps. The source based interleaverldeinterleaver is used for the proposed unequal sample protection. R!S code encodeddecoder uses the RS code to protect the MPEG frame. Lastly, the block interleaver and deinterleaver reshuffle the sequence of the symbol in order to minimize to the effect on the data due to burst errors.
important samples are protected more. For each sample, an error in the MSB generally creates a much more audible distortion than an error in the LSB. For example, a 6-bit data sample 1111 I1 equals to 127 in decimal. If an error occurs in the LSB, the received bit pattern becomes 111110 and the received sample equals to 126 in decimal. The difference is only 0.787%. However, if error occurs in the MSB, the received bit pattern becomes 011111 and the sample equals to 63 in decimal. Tlhe distortion created is 50.39% of the value of the samplle. Therefore, we should use more code bits to protect the MSB rather than the LSB. In our proposed scheme, we take into account the fact that different samples have different perceptual importance as well as the fact that, for each sample, the MSB is more important than the LSB. During the MPEG encoding process of MPEG/Audio, tlhe number of bits allocated for each data sample is known. This information is sent to the RS encoder to determine the level of importance of different bits in each data sample. Bits from different samples are grouped together according to their importance in an interleaver. As the size and the configuration of the interleaver depends on the bit allocation obtain in the MPEG encoder, we call this interleaver the "source-based'' interleaver. Figure 1 shows the block diagram of the unequal sample protection scheme.
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MPEGJ Audio
Encoder
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MPEG Data
Source Based Interleaver
--b RS Coded
4. UNEQUAL ERROR PROTECTION SCHIEME Among all the bits in a MPEG frame, some are more important than the others with respect to the quality of the audio signal. Since the coding resource, i.e. the number of coding bits is fixed, it is better to allocate more resource to those bits that are more relevant to the quality of the decoded audio. Here we propose a new coding scheme based on the unequal error protection idea. The unequal error protection scheme has two parts. The first part is called unequal frame protection. In the MPEG audio frame, the header portion is more important than the sample part since it contains the bits allocation information, SCSFI and scale factors. Without this information, the decoder cannon reconstruct the audio signal back to the 32 equal-width frequency subbainds even with the correct data samples. As the frame has to be (discarded when error occurs in the header section, more protection is needed for the header part. Although the number of bits of the header portion in the MPEG/Audio frame is not fixed, the variation is very small. We assume the first one-third of the frame is the header and the remaining two-third is the data sample portion. In our simulation, a (63, 39) RS code is used to code the header portion and a (63, 45) RS code is used for the data sample portion.
Number of bits of each sample Fig. 1 Block Diagram of Sub-sample unequal protection scheme The data samples entering the "source-based'' interleaver is storied row-by-row from top to bottom with one sample per row. Samples with different bit width are aligned by their LSBs with the leading space in each row marked by "X". The output bits aire read out column-by-column from left to right and from top-tobottom within each column. In the read out process, all1 the spaces marked by "X" are skipped over. For example, we have 4 samples a, b, c and d which aire quantized into 5, 7, 4, 8 bits respectively. Figure 2 shows the content of the source-based interleaver for this example. The bit data is read out column by column It is clear from this example that after interleaving, the MSBs of the perceptually more important samples are grouped together in the front of the sequence while the LSBs of the samples are group in the tail of the sequence.
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The second part of our proposed scheme is called unequal sample protection. In a MPEC/Audio frame, data samples from different subbands are quantized using different number of Ibits. The MPEG/Audio encoder allocates different number OF bits to represent each sample in different subbands, taken into account the psychoacoustic effect. Hence, the resulted bit allocation indicates the relative perceptual importance of the samples. In our unequal sample protection scheme, the more perceptually
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Fig.2 Source-based interleaver
The source-based interleaver transforms the temporal stream of MPEG samples within each frame into a stream of bits with gradually decreasing importance. In other words, the bits at the head of the streams are more important than the bits at the tail of the streams. Therefore, different parts of the stream can be protected using different code rate. In this work, we divide the sample stream into three parts and protected them unequally. A (63,41) RS code is used to encode the first part or the most important bits. A (63, 45) RS code and a (63,49) RS code are used to encode the second and the third parts of the stream, respectively. We have purposely select the code rate such that the overall code rate of the stream is the same as that in equal error protection.
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5. SIMULATION RESULT I
5.1 MEASURING METHOD Since there is no globally recognized and standardized objective method in measuring the quality of audio data, we define some measurement parameters to compare different protection schemes. For the unequal frame protection, we use the frame error rate (FER) to measure the performance of the different protection scheme. For the unequal sample protection, we use two methods to measure the performance. The first is the perceptually weighted mean square error (MSE) of the received frame data. For the second method, we use SMROrlglndl to denote the signal-to-mask ratio in each subband of the decoded audio when the channel is a noiseless channel. We also use SMR,,,,, (SMRunequal) to denote the signal-to-mask ratio in each subband of the decoded audio when the channel suffers from interference and when the system employs equal (unequal) sample protection scheme. Then, we use the subband-error-power-to-mask ratio, defined as the difference between the SMReqUal(SMR,,,,,,J and the SMRO,,~,,~~, for comparison. This difference measures whether the error in each subband is significant when compared to the masking threshold of that subband. Even though the error may be large in absolute value, but if it is relatively small when compared to the masking threshold of that subband, the error does not have significant effect on the audio quality. Furthermore, there will be a larger chance that the error is audible if the error-to-mask ratio is high; therefore, we use the peak subband-error-power-to-mask ratio to compare the quality of the decoded audio with error.
Fig. 3 Comparison of performance between equal frame protection and unequal frame protection at various percentage of interference The results are shown in Figure 3. It indicates that when 10% of the band suffers from interference, the unequal frame protection scheme reduces the frame error rate by a factor of 2.
5.4 SIMULATION RESULT OF UNEQUAL SAMPLE PROTECTION We also compare the performance of systems with both unequal frame protection and unequal sample protection and systems with only unequal frame protection. In Figure 4, we show the mean square error of the MPEG encoded data samples for the two systems under various percentages of interference. As we want to show the effectiveness of the unequal sample protection scheme, we do not count the sample errors in frames that have error in their header part. It is shown in Figure 4 that the mean square sample error increases with the interference level. In some cases, the MSE decreases due to the fact that more frames have been dropped at those particular interference levels. Figure 4 also shows that under all cases, there is a significant improvement (more than IOdB) in the mean square sample error when unequal sample protection is used.
5.2 BENCHMARK MUSIC The following three pieces of music of different styles have been simulated. Each piece is about 30 seconds long. 1.
2.
Classical Music (Gloria RV 589 in D Major - Antonio Vivaldi) Soft Pop Song (That's why you go away - Michael Leans to Rock)
3.
Rock Pop Song (D.I.S.C.0 (Extended Remix) - N-Trance)
5.3 SIMULATION RESULT FOR UNEQUAL FRAME
PROTECTION In this simulation, we try two different frame protection schemes on music #3: equal frame protection (RS(63, 43) for the whole frame) and unequal frame protection (header: RS(63,39), sample: RS(63,45)). In both cases, the total coded bit rate is 256kbps and the coded frames are transmitted over a simulated channel with various percentage of interference.
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Fig. 4 Comparison of the sample mean square error on different music under different schemes
1. 2-3.
4.
% interference Peak-subband-error-power-to-mask of the audio data with/without unequal sample protection Improvement in db
Fig. 5 Difference between the received audio signal under the environment with 15% interference
Table 8. Summary of peak-subband-error power-to-mask for different music.
Using the peak subband-error-power-to-mask ratiio as the performance measure, we have simulated the systems under three different interference levels. In Figure 5, 6, and 7, we plot the number of times that the subband-error-power-to-mask ratio is larger than xdB versus x when the interference level is 15%, lo%, and 6%, respectively.
Table 8 and 9 summarize the peak and average sulbband-errorpower-to-mask ratios under different interference levels. For the peak subband-error-power-to-mask ratio, except the case of the classical music with 6% interference, the improveiment ranges from 2dB to 1ldB in various environment. As for the average peak-subband-error-power-to-mask ratio, the improvemlent achieved by the proposed scheme under different music :and interference levels ranges from 2.83dB to 6.17dB.
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15% 10% 6%
6.17 dB 5 17dR
Fig. 9 Summary of improvement in peak-subband-error-powerto-mask of the audio data
6. CONCLUSIONS Fig. 6 Difference between the received audio signal under the environment with 10% interference 7 L
Two unequal error protection schemes for wireless transmission of MPEG audio were proposed and simulated. By various methods of measurement, it is found that the quality of the audio data is improved after employing these two schemes in the digital wireless audio environment.
7. REFERENCES 1.
2. 3.
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Fig. 7 Difference between the received audio signal the environment with 6% interference
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P.Hoeher, H.Schulze "Performance of an R.CPC-Coded OFDM-based Digital Audio Broadcasting (DAB) System", GLOBEM '-91. Wicker Bhargava, "Reed-Solomon Codes and their applications", IEEE Press 1994. ISO/IEC International Standard IS: 11 172-3 "Information-Technology : Coding of Moving Pictures and Associated Audio for Digital Storage Media at up to about ISMbits/s : Part 3 Audio." J.H. Rothweiler, "Polyphase Quadrature Filters - A New Subband Coding Technique." Proc. I[nternational Conf. IEEE ASSP, 27.2 IEEE Press, Piscabtaway,N.J., 1983, pp. 1280-1283. Davis Pan, "A tutorial on MPEG/Audio Compression", Motorola 1991.
