Hybrid Codecs

Hybrid codecs attempt to fill the gap between waveform and source codecs. As described above waveform coders are capable of providing good quality speech at bit rates down to about 16 kbits/s, but are of limited use at rates below this. Vocoders on the other hand can provide intelligible speech at 2.4 kbits/s and below, but cannot provide natural sounding speech at any bit rate. Although other forms of hybrid codecs exist, the most successful and commonly used are time domain Analysis-by-Synthesis (AbS) codecs. Such coders use the same linear prediction filter model of the vocal tract as found in LPC vocoders. However instead of applying a simple two-state, voiced/unvoiced, model to find the necessary input to this filter, the excitation signal is chosen by attempting to match the reconstructed speech waveform as closely as possible to the original speech waveform. AbS codecs were first introduced in 1982 by Atal and Remde with what was to become known as the Multi-Pulse Excited (MPE) codec. Later the Regular-Pulse Excited (RPE), and the Code-Excited Linear Predictive (CELP) codecs were introduced. These coders will be discussed briefly here.

A general model for AbS codecs is shown in Figure 6.

Figure 6: AbS Codec Structure

AbS codecs work by splitting the input speech to be coded into frames, typically about 20 ms long. For each frame parameters are determined for a synthesis filter, and then the excitation to this filter is determined. This is done by finding the excitation signal which when passed into the given synthesis filter minimes the error between the input speech and the reconstructed speech. Thus the name Analysis-by-Synthesis - the encoder analyses the input speech by synthesising many different approximations to it. Finally for each frame the encoder transmits information representing the synthesis filter parameters and the excitation to the decoder, and at the decoder the given excitation is passed through the synthesis filter to give the reconstructed speech.

The synthesis filter is usually an all pole, short-term, linear filter of the form


is the prediction error filter determined by minimising the energy of the residual signal produced when the original speech segment is passed through it. The order p of the filter is typically around ten. This filter is intended to model the correlations introduced into the speech by the action of the vocal tract.

The synthesis filter may also include a pitch filter to model the long-term periodicities present in voiced speech. Alternatively these long-term periodicities may be exploited by including an adaptive codebook in the excitation generator so that the excitation signal u(n) includes a component of the form Gu(n-), where is the estimated pitch period. Generally MPE and RPE codecs will work without a pitch filter, although their performance will be improved if one is included. For CELP codecs however a pitch filter is extremely important, for reasons discussed below.

The error weighting block is used to shape the spectrum of the error signal in order to reduce the subjective loudness of this error. This is possible because the error signal in frequency regions where the speech has high energy will be at least partially masked by the speech. The weighting filter emphasises the noise in the frequency regions where the speech content is low. Thus minimising the weighted error concentrates the energy of the error signal in frequency regions where the speech has high energy. Therefore the error signal will be at least partially masked by the speech, and so its subjective importance will be reduced. Such weighting is found to produce a significant improvement in the subjective quality of the reconstructed speech for AbS codecs.

The distinguishing feature of AbS codecs is how the excitation waveform u(n) for the synthesis filter is chosen. Conceptually every possible waveform is passed through the filter to see what reconstructed speech signal this excitation would produce. The excitation which gives the minimum weighted error between the original and the reconstructed speech is then chosen by the encoder and used to drive the synthesis filter at the decoder. It is this `closed-loop' determination of the excitation which allows AbS codecs to produce good quality speech at low bit rates. However the numerical complexity involved in passing every possible excitation signal through the synthesis filter is huge. Usually some means of reducing this complexity, without compromising the performance of the codec too badly, must be found.

The differences between MPE, RPE and CELP codecs arise from the representation of the excitation signal u(n) used. In multi-pulse codecs u(n) is given by a fixed number of non-zero pulses for every frame of speech. The positions of these non-zero pulses within the frame, and their amplitudes, must be determined by the encoder and transmitted to the decoder. In theory it would be possible to find the very best values for all the pulse positions and amplitudes, but this is not practical due to the excessive complexity it would entail. In practice some sub-optimal method of finding the pulse positions and amplitudes must be used. Typically about 4 pulses per 5 ms are used, and this leads to good quality reconstructed speech at a bit-rate of around 10 kbits/s.

Like the MPE codec the Regular Pulse Excited (RPE) codec uses a number of non-zero pulses to give the excitation signal u(n). However in RPE codecs the pulses are regularly spaced at some fixed interval, and the encoder needs only to determine the position of the first pulse and the amplitude of all the pulses. Therefore less information needs to be transmitted about pulse positions, and so for a given bit rate the RPE codec can use many more non-zero pulses than MPE codecs. For example at a bit rate of about 10 kbits/s around 10 pulses per 5 ms can be used in RPE codecs, compared to 4 pulses for MPE codecs. This allows RPE codecs to give slightly better quality reconstructed speech quality than MPE codecs. However they also tend to be more complex. The pan-European GSM mobile telephone system uses a simplified RPE codec, with long-term prediction, operating at 13 kbits/s to provide toll quality speech.

Although MPE and RPE codecs can provide good quality speech at rates of around 10 kbits/s and higher, they are not suitable for rates much below this. This is due to the large amount of information that must be transmitted about the excitation pulses' positions and amplitudes. If we attempt to reduce the bit rate by using fewer pulses, or coarsely quantizing their amplitudes, the reconstructed speech quality deteriorates rapidly. Currently the most commonly used algorithm for producing good quality speech at rates below 10 kbits/s is Code Excited Linear Prediction (CELP). This approach was proposed by Schroeder and Atal in 1985, and differs from MPE and RPE in that the excitation signal is effectively vector quantized. The excitation is given by an entry from a large vector quantizer codebook, and a gain term to control its power. Typically the codebook index is represented with about 10 bits (to give a codebook size of 1024 entries) and the gain is coded with about 5 bits. Thus the bit rate necessary to transmit the excitation information is greatly reduced - around 15 bits compared to the 47 bits used for example in the GSM RPE codec.

Originally the codebook used in CELP codecs contained white Gaussian sequences. This was because it was assumed that long and short-term predictors would be able to remove nearly all the redundancy from the speech signal to produce a random noise-like residual. Also it was shown that the short-term probability density function (pdf) of this residual was nearly Gaussian. Schroeder and Atal found that using such a codebook to produce the excitation for long and short-term synthesis filters could produce high quality speech. However to choose which codebook entry to use in an analysis-by-synthesis procedure meant that every excitation sequence had to be passed through the synthesis filters to see how close the reconstructed speech it produced would be to the original. This meant the complexity of the original CELP codec was much too high for it to be implemented in real-time - it took 125 seconds of Cray-1 CPU time to process 1 second of the speech signal. Since 1985 much work on reducing the complexity of CELP codecs, mainly through altering the structure of the codebook, has been done. Also large advances have been made with the speed possible from DSP chips, so that now it is relatively easy to implement a real-time CELP codec on a single, low cost, DSP chip. Several important speech coding standards have been defined based on the CELP principle, for example the American Department of Defence (DoD) 4.8 kbits/s codec, and the CCITT low-delay 16 kbits/s codec .

The CELP coding principle has been very successful in producing communications to toll quality speech at bit rates between 4.8 and 16 kbits/s. The CCITT standard 16 kbits/s codec produces speech which is almost indistinguishable from 64 kbits/s log-PCM coded speech, while the DoD 4.8 kbits/s codec gives good communications quality speech. Recently much research has been done on codecs operation below 4.8 kbits/s, with the aim being to produce a codec at 2.4 or 3.6 kbits/s with speech quality equivalent to the 4.8 kbits/s DoD CELP. We will briefly describe here a few of the approaches which seem promising in the search for such a codec.

The CELP codec structure can be improved and used at rates below 4.8 kbits/s by classifying speech segments into one of a number of types (for example voiced, unvoiced and transition frames). The different speech segment types are then coded differently with a specially designed encoder for each type. For example for unvoiced frames the encoder will not use any long-term prediction, whereas for voiced frames such prediction is vital but the fixed codebook may be less important. Such class-dependent codecs have been shown to be capable of producing reasonable quality speech at rates down to 2.4 kbits/s. Multi-Band Excitation (MBE) codecs work by declaring some regions in the frequency domain as voiced and others as unvoiced. They transmit for each frame a pitch period, spectral magnitude and phase information, and voiced/unvoiced decisions for the harmonics of the fundamental frequency. Originally it was shown that such a structure was capable of producing good quality speech at 8 kbits/s, and since then this rate has been significantly reduced. Finally Kleijn has suggested an approach for coding voiced segments of speech called Prototype Waveform Interpolation (PWI). This works by sending information about a single pitch cycle every 20-30 ms, and using interpolation to reproduce a smoothly varying quasi-periodic waveform for voiced speech segments. Excellent quality reproduced speech can be obtained for voiced speech at rates as low as 3 kbits/s. Such a codec can be combined with a CELP type codec for the unvoiced segments to give good quality speech at rates below 4 kbits/s.

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