Data processing: speech signal processing – linguistics – language – Speech signal processing – For storage or transmission
Reexamination Certificate
1999-09-29
2002-09-17
Dorvil, Richemond (Department: 2641)
Data processing: speech signal processing, linguistics, language
Speech signal processing
For storage or transmission
C704S201000, C704S261000, C704S265000
Reexamination Certificate
active
06453287
ABSTRACT:
FIELD OF THE INVENTION
The present invention relates to speech signal coding using a parametric coder to model a speech waveform. The speech signal parameters are communicated via a communications channel and used to synthesize the speech waveform at the receiver. More specifically, the present invention enhances the speech quality and reduces the computations of the mixed excitation linear predictive (MELP) speech coder.
BACKGROUND OF THE INVENTION
Low bit-rate speech coding technology is widely used for digital voice communication in narrow-bandwidth channels. The common objective of this technology is to transfer the digital speech signal information at a low bit rate (typically 2,400 bits/sec) while providing good quality speech synthesis at the destination. This technology also strives to provide low computational complexity, low memory requirements, and a small algorithmic delay particularly for real-time low-cost voice communications.
FIG. 1A
illustrates the general environment surrounding speech encoders and decoders as used in a one-way communications system. Full duplex communications are easily enabled by integrating both an encoder and decoder at both ends of the communications system.
The first widely used low bit-rate speech coder was the Federal Standard linear predictive coding (LPC) vocoder (FS1015) in which either a periodic pulse train or white noise excites an all-pole filter in order to synthesize speech. While the 2.4 kbps bit rate was attractive, the LPC vocoder was not acceptable for many speech applications as users characterized the synthesized speech as synthetic and buzzy.
The LPC vocoder analyzes the speech waveform and extracts such parameters as filter coefficients, pitch period, voicing decision, and gain are updated every 20-30 ms and transmitted to the communications channel. The artifacts residing in the traditional LPC vocoder include buzzes, clicks, and tonal noise. In addition, the speech quality is very poor in the presence of background noise. These unintended additions to the synthesized speech are the result of the simple excitation model and the binary voicing decision error.
Over the years, several low bit-rate speech coding algorithms have been developed, and some state-of-the-art coders now provide a good natural quality. The mixed excitation linear predictive (MELP) coder is one of these speech coders. The MELP coder is a linear-prediction-based speech coder includes five features not found in the LPC vocoder: mixed excitation, aperiodic pulses, adaptive spectral enhancement, pulse dispersion, and Fourier magnitude modeling. These features improve the synthesized speech quality by removing distortions resident in the LPC vocoder. FIG.
1
B and
FIG. 1C
illustrate block diagrams of the MELP encoder and decoder respectively.
However, the MELP still has some perceivable distortions, particularly around the non-stationary speech segments and for some low-pitch male speakers. These distortions can also be observed with other low bit-rate speech coders. The distortion around the non-stationary speech segments results from the update of speech parameters at a low frame rate (typically 30-50 frames/sec). It is known that increasing the frame rate helps to solve this problem. Unfortunately, this solution requires a much higher bit rate. Another possible solution is a variable frame-rate system that updates the speech parameters in the less stationary segments at a higher frame rate while maintaining a low frame rate in the stationary segments. Such an approach is provided by the delayed decision approach based on dynamic programming, which uses the future frame information to control the frame rate. This system can produce high-quality speech while maintaining a relatively low bit rate by reducing the average frame rate. However, this method requires a considerably longer algorithmic delay (around 150 ms), which is unacceptable in many applications (such as two-way voice communications).
The distortion for low-pitch male speakers in the MELP is characterized by a high-pass filtered quality of the coded speech. In other words, the synthesized speech lacks “sound pressure” in the low frequencies. This distortion is caused by a post filter and a preprocessing high-pass filter, which are used in the modern low bit-rate speech coders to remove 60 Hz noise and to enhance the coded speech quality. These filters suppress the harmonic magnitudes in the low frequencies, particularly for low-pitch male speakers whose fundamental frequencies are less than 100 Hz. The suppression of these low frequency harmonics results in a high-pass filtered speech that is perceived as too synthetic.
The most significant speech distortion present in the prior art is the lack of a suitable model or method to accurately synthesize a plosive sound. Plosive sounds are characterized by the sudden opening or closing of the vocal chords. Plosive phonemes are created when most English speaking persons create sounds such as:“b,” “d,” “g,” “k,” “p,” “t,” “th,” “ch,” or “tch.” It is important to note that the preceding list of plosive phonemes is not exclusive and that not all speakers will create like sounds. Plosive phonemes may be created both at the start and at the end of syllables (i.e. “pop,” “tank,” “tot”), at the end of syllables (i.e. “sound,” “sat”, “shrug”) or at the start of syllables (i.e. “toy.” “boy,” “boss”). Plosive sounds are easily identified in a speech waveform but difficult to model and synthesize in low bit-rate speech coders. Plosive sounds are characterized by an impulse of energy followed by a brief period where the speech waveform is aperiodic. Prior art speech encoders have been unable to model and synthesize plosive sounds in a manner acceptable to the human ear.
SUMMARY OF THE INVENTION
As described briefly, an object of the present invention is to enhance the coded speech quality of the existing low bit rate speech coders including the MELP vocoder while maintaining its low bit rate, small algorithmic delay, and low computational complexity.
Another object of the present invention is to provide an efficient mixed excitation algorithm to reduce the computational complexity of the existing MELP vocoder. Another object of the present invention is to provide bit-stream compatibility with the existing MELP vocoder in order to permit the introduction of the invention into systems where only the present MELP decoder is available. This would allow for backward compatibility through the introduction of an updated encoder while allowing for full system upgrades where both the encoder and the decoder could be updated.
The present invention provides four embodiments. The first is a robust pitch detection algorithm. In the encoder, the fixed-length pitch analysis window is manipulated around the original position to seek the position that contains the signal with the highest pitch correlation. Once the window position is determined, pitch is estimated using the signal that is contained in the selected window. Other parameters such as LPC coefficients, gain, and voicing decision are also estimated using the signal corresponding to the selected window. The estimated parameters are used to synthesize the coded speech in the decoder on each sample window in the same manner as earlier fixed-position windows in the prior art.
The second embodiment is a plosive analysis/synthesis method. In the encoder, the system first detects the frame that contains the plosive signal. The plosive detection is performed with sliding-window peakiness analysis. The detected plosive signal is quantized to only a small number of bits and transmitted via the communication channel to the decoder. In the decoder, the plosive signal is synthesized independently and added back to the coded speech.
The third embodiment is a post processor for the Fourier magnitude model. In the decoder, the harmonic magnitudes of the coded speech in the low frequencies are emphasized to overcome the muffling effect of the high pass filter. In this way, the decoded speech is synthesized without the muffl
Barnwell, III Thomas P.
Truong Kwan K.
Unno Takahiro
Dorvil Richemond
Georgia-Tech Research Corporation
Nolan Daniel
Thomas Kayden Horstemeyer & Risley LLP
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