Lost frame recovery techniques for parametric, LPC-based...

Details Lost frame recovery techniques for parametric, LPC-based... Lost frame recovery techniques for parametric, LPC-based...

1998-06-19

2004-10-26

McFadden, Susan (Department: 2655)

Data processing: speech signal processing, linguistics, language

Speech signal processing

For storage or transmission

C704S214000

Reexamination Certificate

active

06810377

ABSTRACT:

BACKGROUND OF THE INVENTION
The transmission of compressed speech over packet-switching and mobile communications networks involves two major systems. The source speech system encodes the speech signal on a frame by frame basis, packetizes the compressed speech into bytes of information, or packets, and sends these packets over the network. Upon reaching the destination speech system, the bytes of information are unpacketized into frames and decoded. The G.723.1, dual rate speech coder, described in
ITU
-
T Recommendation G.
723.1, “Dual Rate Speech Coder for Multimedia Communications Transmitting at 5.3 and 6.3 kbit/s,” March 1996 (hereafter “Reference 1”, and incorporated herein by reference) was ratified by the ITU-T in 1996 and has since been used to add voice over various packet-switching as well as mobile communications networks. With a mean opinion score of 3.98 out of 5.0 (see, Thryft, A. R., “Voice over IP Looms for Intranets in '98,”
Electronic Engineering Times
, August, 1997, Issue: 967, pp. 79, 102, hereafter “Reference
2
”, and incorporated herein by reference), the near toll quality of the G.723.1 standard is ideal for real-time multimedia applications over private and local area networks (LANs) where packet loss is minimal. However, over wide area networks (WANs), global area networks (GANs), and mobile communications networks, congestion can be severe, and packet loss may result in heavily degraded speech if left untreated. It is therefore necessary, to develop techniques to reconstruct lost speech frames at the receiver in order to minimize distortion and maintain output intelligibility.
The following discussion of the G.273.1 dual rate coder and its error concealment will assist in a full understanding of the invention.
The G.723.1 dual rate speech coder encodes 16-bit linear pulse-code modulated (PCM) speech, sampled at a rate of 8 KHz, using linear predictive analysis-by-synthesis coding. The excitation for the high rate coder is Multipulse Maximum Likelihood Quantization (MP-MLQ) while the excitation for the low rate coder is Algebraic-Code-Excited Linear-Prediction (ACELP). The encoder operates on a 30 ms frame size, equivalent to a frame length of 240 samples, and divides every frame into four subframes of 60 samples each. For every 30 ms speech frame, a 10th order Linear Prediction Coding (LPC) filter is computed and its coefficients are quantized in the form of Line Spectral Pair (LSP) parameters for transmission to the decoder. An adaptive codebook pitch lag and pitch gain are then calculated for every subframe and transmitted to the decoder. Finally, the excitation signal, consisting of the fixed codebook gain, pulse positions, pulse signs, and grid index, is approximated using either MP-MLQ for the high rate coder or ACELP for the low rate coder, and transmitted to the decoder. In sum, the resulting bitstream sent from encoder to decoder consists of the LSP parameters, adaptive codebook lags, fixed and adaptive codebook gains, pulse positions, pulse signs, and the grid index.
At the decoder, the LSP parameters are decoded and the LPC synthesis filter generates reconstructed speech. For every subframe, the fixed and adaptive codebook contributions are sent to a pitch postfilter, whose output is input to the LPC synthesis filter. The output of the synthesis filter is then sent to a formant postfilter and gain scaling unit to generate the synthesized output. In the case of indicated frame erasures, an error concealment strategy, described in the following subsection, is provided.
FIG. 1
displays a block diagram of the G.723.1 decoder.
In the presence packet of losses, current G.723.1 error concealment involves two major steps. The first step is LSP vector recovery and the second step is excitation recovery. In the first step, the missing frame's LSP vector is recovered by applying a fixed linear predictor to the previously decoded LSP vector. In the second step, the missing frame's excitation is recovered using only the recent information available at the decoder. This is achieved by first determining the previous frame's voiced/unvoiced classifier using a cross-correlation maximization function and then testing the prediction gain for the best vector. If the gain is more than 0.58 dB, the frame is declared as voiced, otherwise, the frame is declared as unvoiced. The classifier then returns a value of 0 if the previous frame is unvoiced, or the estimated pitch lag if the previous frame is voiced. In the unvoiced case, the missing frame's excitation is then generated using a uniform random number generator and scaled by the average of the gains for subframes
2
and
3
of the previous frame. Otherwise, for the voiced case, the previous frame is attenuated by 2.5 dB and regenerated with a periodic excitation having a period equal to the estimated pitch lag. If packet losses continue for the next two frames, the regenerated excitation is attenuated by an additional 2.5 dB for each frame, but after three interpolated frames, the output is completely muted, as described in Reference 1.
The G.723.1 error concealment strategy was tested by sending various speech segments over a network with packet loss levels of 1%, 3%, 6%, 10%, and 15%. Single as well as multiple packet losses were simulated for each level. Through a series of informal listening tests, it was shown that although the overall output quality was very good for lower levels of packet loss, a number of problems persisted at all levels and became increasingly severe as packet loss increased.
First, parts of the output segment sounded unnatural and contained many annoying, metallic-sounding artifacts. The unnatural sounding quality of the output can be attributed to LSP vector recovery based on a fixed predictor as previously described. Since the missing frame's LSP vector is recovered by applying a fixed predictor to the previous frame's LSP vector, the spectral changes between the previous and reconstructed frames are not smooth. As a result of the failure to generate smooth spectral changes across missing frames, unnatural sounding output quality occurs, which increases unintelligibility during high levels of packet loss. In addition, many high-frequency, metallic-sounding artifacts were heard in the output. These metallic-sounding artifacts primarily occur in unvoiced regions of the output, and are caused by incorrect voicing estimation of the previous frame during excitation recovery. In other words, since a missing, unvoiced frame may incorrectly be classified as voiced, then transition into the missing frame will generate a high-frequency glitch, or metallic-sounding artifact, by applying the estimated pitch lag computed for the previous frame. As packet loss increases, this problem becomes even more severe, as incorrect voicing estimation generates increased distortion.
Another problem using G.723.1 error concealment was the presence of high-energy spikes in the output. These high-energy spikes, which are especially uncomfortable for the ear, are caused by incorrect estimation of the LPC coefficients during formant postfiltering, due to poor prediction of the LSP or gain parameter, using G.723.1 fixed LSP prediction and excitation recovery. Once again, as packet loss increases, the number of high-energy spikes also increases, leading to greater listener discomfort and distortion.
Finally, “choppy” speech, resulting from complete muting of the output, was evident. Since G.723.1 error concealment reconstructs no more than three consecutive missing frames, all remaining missing frames are simply muted, leading to patches of silence in the output, or “choppy” speech. Since there is a greater probability that more than three consecutive packets may be lost in a network, when packet loss increases, this will lead to increased “choppy” speech and hence, decreased intelligibility and distortion at the output.
SUMMARY OF THE INVENTION
It is an object of the present invention to eliminate the above problems and improve upon the error concealment strategy defi

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