Classifications

• • G—PHYSICS
  • G10—MUSICAL INSTRUMENTS; ACOUSTICS
  • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
  • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
  • G10L19/02—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using spectral analysis, e.g. transform vocoders or subband vocoders
• • G—PHYSICS
  • G10—MUSICAL INSTRUMENTS; ACOUSTICS
  • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
  • G10L21/00—Speech or voice signal processing techniques to produce another audible or non-audible signal, e.g. visual or tactile, in order to modify its quality or its intelligibility
  • G10L21/02—Speech enhancement, e.g. noise reduction or echo cancellation
  • G10L21/038—Speech enhancement, e.g. noise reduction or echo cancellation using band spreading techniques
• • G—PHYSICS
  • G10—MUSICAL INSTRUMENTS; ACOUSTICS
  • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
  • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
  • G10L19/04—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using predictive techniques
  • G10L19/06—Determination or coding of the spectral characteristics, e.g. of the short-term prediction coefficients
• • G—PHYSICS
  • G10—MUSICAL INSTRUMENTS; ACOUSTICS
  • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
  • G10L21/00—Speech or voice signal processing techniques to produce another audible or non-audible signal, e.g. visual or tactile, in order to modify its quality or its intelligibility
  • G10L21/02—Speech enhancement, e.g. noise reduction or echo cancellation

Definitions

• the invention relates to a method and a device for artificially expanding the bandwidth of speech signals.
• Speech signals cover a wide frequency range, which ranges from the basic speech frequency, which is dependent on the speaker in the range between 80 to 160 Hz, to the frequencies beyond 10 kHz.
• the basic speech frequency which is dependent on the speaker in the range between 80 to 160 Hz
• the frequencies beyond 10 kHz for voice communication over certain transmission media, such as telephones, only a limited portion is transmitted for bandwidth efficiency, ensuring a sentence intelligibility of about 98%.
• a speech signal can essentially be subdivided into three frequency ranges.
• Each of these frequency ranges characterizes specific speech characteristics as well as subjective sensations. This results in lower frequencies below about 300 Hz, essentially during voiced speech segments, such as vowels.
• This frequency range in this case contains tonal components, i. H. in particular the basic voice frequency and, depending on the pitch, possibly some harmonics.
• these depth frequencies are essential.
• the speech base frequency can be perceived by a human listener due to the psychoacoustic property of the virtual pitch sensation even in the absence of the depth frequencies from the harmonic structure in higher frequency ranges.
• medium frequencies in the range of about 300 Hz to about 3.4 kHz in voice activities are fundamental. additionally available in the voice signal.
• Their time-variant spectral coloring by several formants as well as the temporal and spectral fine structure characterize the respective spoken sound or phoneme.
• the middle frequencies carry the bulk of the information relevant to the intelligibility of the language.
• the speech quality is a subjective quantity with a plurality of components, of which the intelligibility of the speech signal is the most important for such a speech communication system.
• parameters of the broadband model are determined from short segments of a narrowband speech signal using pattern recognition methods, which are then used to estimate the missing signal components of the speech.
• the narrow-band speech signal becomes a broadband equivalent with frequency components in the range 50
• This secondary information tions are transmitted in a coded bit stream to a receiver.
• Other standards based on the extension of the bandwidth technique are currently seen in the AMR-WB + and extended aacPlus speech / audio coding / decoding techniques.
• Methods designed to encode and decode information are referred to as codecs and include both an encoder and a decoder.
• Any digital telephone, whether built for a fixed or mobile network, includes such a codec that converts analog to digital signals and digital to analog. Such a codec can be implemented in hardware or in software.
• components of an extension band for example in the frequency range of 6.4 to 7 kHz, are encoded and decoded using the aforementioned LPC encoding technique.
• an LPC analysis of the extension band of the input signal is performed in an encoder and the LPC coefficients and the amplification factors of subframes of a residual signal are encoded.
• the remainder of the expansion band is generated and the transmitted gain factors and the LPC synthesis filters are used to generate an output signal.
• the procedure described above can be applied either directly to the wideband input signal or else to a subband signal of the extension band that is downsampled in the limit range or in the critical range.
• the extended aacPlus encoding standard uses SBR (Spectral Band Replication) technology.
• the broadband audio signal is split into frequency subbands by means of a 64-channel QMF filter bank.
• a sophisticated and technically advanced parametric coding is applied to the sub-bands of the signal components, requiring and using a large number of detectors and estimators to control the bitstream contents.
• an improvement in speech intelligibility and speech quality in the transmission of speech signals can be achieved, whereby speech signals are also understood as audio signals.
• the inventive method is also very robust against interference during transmission.
• the signal components required for bandwidth expansion are determined by filtering, in particular bandpass filtering, from the wideband input speech signal, whereby a simple and low-cost selection of the required signal components can be performed.
• Determining the temporal envelopes in step c) is preferably carried out independently of the determination of the spectral envelopes in step d). As a result, the determination of the envelope takes place in a precise manner, whereby a mutual influence can be avoided.
• step e prior to encoding the temporal envelope and the spectral envelope in step e), quantization of the temporal envelope and the spectral envelope is performed.
• the signal powers of spectral subbands of the signal components intended for bandwidth expansion are determined in step d) for determining the spectral envelopes. The determination of the characterization of the temporal and the spectral envelope can thereby be carried out very accurately.
• signal segments of the bandwidth tenerweittation certain signal components generated, these signal segments in particular transformed, in particular FF (Fast Fourier) transformed, are.
• the signal powers of temporal signal segments of the signal components intended for bandwidth expansion are advantageously determined in step c) for determining the time-dependent envelopes. In a labor-saving manner, the determination of the required parameters can thereby be carried out.
• step f) the encoded information for reconstructing the temporal envelope and the spectral envelope are decoded.
• An excitation signal is advantageously generated in a decoder from a signal transmitted to the decoder, the transmitted signal having such a signal power in the frequency range which corresponds to that of the extension signal of the wideband input speech signal, which enables generation of an excitation signal.
• a modulated narrowband signal having a band range with frequencies below the frequencies of the band range of the extension band of the wideband input speech signal for generating the excitation signal is preferably transmitted to the decoder.
• the excitation signal preferably has harmonics of the fundamental frequency of the signal transmitted to the decoder.
• a first correction factor is advantageously determined. Furthermore, from the first correction factor and the excitation signal, a reconstructive shaping of the temporal envelope, in particular by a multiplication of the first correction factor with the excitation signal, is performed. In addition, the reconstructed shaping of the temporal envelopes is filtered in an advantageous manner and impulse responses are generated during filtering. From the impulse responses and the reconstructed research tion of the temporal envelope, a reconstructive shaping of the spectral envelope is performed. Furthermore, the signal components of the expansion band of the wideband input speech signal are reconstructed from the reconstructed shaping of the spectral envelope. The reconstruction of the temporal and the spectral envelopes can be carried out very reliably and very accurately.
• a narrowband signal having a band range with frequencies below the frequencies of the extension band of the broadband input signal is transmitted to the decoder.
• the bandwidth-expanded output speech signal is advantageously determined from the narrow-band signal transmitted to the decoder and the reconstructed shaping of the spectral envelope, in particular from a summation of these two signals, and is provided as an output signal of the decoder.
• an output signal can be generated and provided which ensures high speech intelligibility and speech quality.
• the steps a) to e) are preferably carried out in an encoder, which is preferably arranged in a transmitter.
• the encoded information generated in step e) is advantageously transmitted as a digital signal to the decoder.
• At least step f) is preferably performed in a receiver with the decoder located in the receiver.
• all steps a) to f) of the method according to the invention are carried out in a receiver.
• steps a) to e) in the receiver are replaced by an estimation method (to be implemented differently).
• the steps a) to e) can also be carried out separately in a transmitter.
• the wideband input speech signal advantageously comprises a bandwidth between about 50 Hz and about 7 kHz.
• the extension band of the wideband input speech signal preferably comprises the frequency range of about 3.4 kHz to about 7 kHz.
• the narrowband signal comprises a
• Signal range of the wideband input speech signal from about 50 Hz to about 3.4 kHz.
• the inventive device enables improved speech quality and improved speech intelligibility of speech signals during transmission in communication devices, such as mobile devices or ISDN devices.
• the means in a) to d) are advantageously designed as encoders.
• the encoder can be in a transmitter or in a receiver, wherein the decoder is arranged in a receiver.
• Advantageous embodiments of the method according to the invention can, insofar as it is transferable, also be regarded as advantageous embodiments of the device according to the invention.
• FIG. 1 shows an encoder of a device according to the invention
• FIG. 2 shows a decoder of a device according to the invention.
• speech signals also includes audio signals.
• FIG 1 and FIG 2 the same or functionally identical elements are provided with the same reference numerals.
• the encoder 1 shows a schematic block diagram representation of an encoder 1 of a device according to the invention for the artificial extension of the bandwidth of speech signals.
• the coder 1 can be implemented as an algorithm both in hardware and in software.
• the encoder 1 comprises a block 11, which is designed for bandpass filtering of a broadband input speech signal s w ' b (k).
• the encoder 1 comprises a block 12 and a block 13, which are connected to the block 11.
• Block 12 is designed to determine the temporal envelope of the signal components intended for bandwidth expansion, which are determined from an extension band of the wideband input speech signal.
• the block 13 is configured to determine the spectral envelope of the bandwidth expansion signal components determined from the extension band of the wideband input speech signal.
• the block 12 and the block 13 are connected to a block 14, the block 14 for quantizing the temporal envelope and the spectral envelope generated by the blocks 12 and 13, respectively be, is trained.
• FIG. 1 further shows a block 2, which is designed as a bandpass filter and to which the broadband input speech signal s w ' b (k) is applied.
• the block 2 is further connected to a further block 3, wherein the block 3 is formed as a further encoder.
• the encoder 1 and the blocks 2 and 3 are arranged in a first telephone set.
• the broadband input speech signal has a bandwidth of approximately 50 Hz to approximately 7 kHz. According to the invention, as can be seen from the illustration in FIG. 1, this wideband input speech signal s w ' b (k) is applied to the bandpass filter or block 11 of the coder 1.
• the signal components required for bandwidth expansion from the expansion band which in the exemplary embodiment comprises a bandwidth of about 3.4 kHz to about 7 kHz, are determined.
• the signal components required for the bandwidth expansion are characterized by the signal s eb (k) and are transmitted as an output signal of the block 11 to the two blocks 12 and 13.
• the signal s eb (k) is transmitted as an output signal of the block 11 to the two blocks 12 and 13.
• block 12 from this signal s eb (k) the temporal
• the spectral envelope of the signal components which are characterized by the signal s eb (k) is determined in block 13.
• Segmented signal components s eb (k) and transformed these fenestrated signal segments The segmentation of the signal s eb (k) takes place in frames with a length of each of k samples. All subsequent steps and subalgorithms are performed frame by frame.
• Each speech frame eg with 10 ms or 20 ms or 30 ms duration
• the windowed signal segments are then transformed.
• a transformation into the frequency domain is carried out by means of an FFT (Fast Fourier Transform).
• FFT-transformed signal segments are determined according to the following formula 1):
• Nf denotes the FFT length or the frame size
• ⁇ denotes the frame index
• Mf denotes the overlap of the frames of the windowed signal segments.
• W y ( ⁇ ) denotes the window function.
• ⁇ denotes the index of the corresponding subband
• EB ⁇ characterizes that set which contains all FFT interval ranges i with non-zero coefficients in the ⁇ th frequency space window w ⁇ (i).
• the signal powers i y ( ⁇ , ⁇ ) of the subbands according to formula 2) characterize the information of the spectral envelopes which are transmitted to a decoder.
• the determination of the time envelopes in the time period is performed in a manner similar to the determination of the spectral envelopes and is based on short-term windowed ones Segments of the band-pass filtered wideband input speech signal s w ' b (k).
• signal segments of the signal s eb (k) are also taken into account in the determination of the time envelopes.
• the signal power is calculated according to formula 3) below:
• N t denotes the frame length
• v denotes the frame index
• M t again denotes the overlap of the frames of the signal segments. It should be noted that, in general, the frame length N t and the overlap of the frames M t used to extract the temporal envelopes are smaller and much smaller than the corresponding magnitudes Nf and Mf, respectively Spectral envelopes are used.
• Short segment signal powers of the filtered parts and the original parts of the signal s eb (k) gives the short time envelope, which is downsampled to determine the signal powers P t (y).
• P t (y) of the signal segments then characterize the temporal envelope information.
• the signals s p (y ⁇ or ⁇ ( ⁇ ) characterizing the temporal envelope and the spectral envelope, which characterize the extracted parameters of the signal powers according to formulas 2) and 3), are quantized and coded in block 14.
• the output signal of the block 14 is a digital signal BWE, which characterizes a bit stream which contains in coded form information of the temporal envelope and the spectral envelope.
• This digital signal BWE is transmitted to a decoder, which will be explained in more detail below. It should be noted that in the case of a redundancy between the extracted parameters of the signal strengths according to formulas 2) and 3), a common coding, such as may be made possible, for example, by vector quantization, can be carried out.
• the wideband input speech signal s w ' b (k) is also transmitted to the block 2.
• this block 2 designed as a bandpass filter, the signal components of a narrowband range of the wideband input speech signal s w ' b (k) are filtered.
• the narrowband range in the exemplary embodiment is between 50 Hz and 3.4 kHz.
• the output signal of the block 2 is a narrowband signal s nb (k) and is transmitted to the block 3, which is formed in the embodiment as a further encoder.
• the narrowband signal s nb (k) is encoded and transmitted as a digital signal BWN as a bit stream to the decoder explained below.
• FIG. 2 shows a schematic block diagram illustration of such a decoder 5 of a device according to the invention for artificially expanding the bandwidth of speech signals.
• the digital signal BWN is first transmitted to a further decoder 4 which decodes the information contained in the digital signal BWN and in turn generates the narrowband signal s nb (k) from it.
• the decoder 4 generates a further signal s s ⁇ (k), which contains side information.
• These side information may be, for example, gain factors or filter coefficients.
• This signal s s ⁇ (k) of the decoder 5 to block wear exceeds 51st
• the block 51 is formed in the embodiment for generating an excitation signal in the frequency range of the extension band, to which the information of the signal s s ⁇ (k) are taken into account.
• the decoder 5 which is arranged in the embodiment in a receiver, a block 52, which is designed for decoding the transmitted over a transmission distance between the encoder 1 and the decoder 2 signal BWE. It should be noted that also the digital signal BWN is transmitted via this transmission path between the encoder 1 and the decoder 5.
• both the block 51 and the block 52 are connected to decoder areas 53 to 55.
• the functional principle of the decoder 5 or the sub-steps of the method according to the invention carried out in the decoder 5 are explained in more detail below.
• the information contained in the encoded digital signal BWE is decoded in block 52 and the signal powers, which are calculated according to formulas 2) and 3) and which characterize the temporal envelope and the spectral envelope, are reconstructed.
• the excitation signal S ⁇ x (Jc) generated in block 51 is the
• This excitation signal can essentially be any excitation signal
• this excitation signal must be, as an essential condition for this signal must be that it has a sufficient signal power in the frequency range of the extension band of the wideband input spectral signal s w ' b (k).
• the narrowband signal s nb (k) or any noise are used as excitation signal s exc (k) is a modulated version.
• this excitation signal is responsible for the fine structuring of the spectral envelope and the temporal envelope in the signal components of the extension band of a broadband output speech signal s wb (k). For this reason, it is advantageous for this excitation signal s (k) to be present in one of these is generated such that it has the harmonics of the fundamental frequency of the narrowband signal s nb (k).
• harmonic frequency excitation is at an integer multiple of the current fundamental frequency by LTP synthesis filtering a bandpass filter (frequency range of the extension band) from an arbitrary signal n eb (k) possible.
• the LTP amplification factor can be reduced or limited by the function f (b) in order to be able to prevent overstimulation of the generated signal components of the expansion band. It should be noted that a plurality of further alternatives can be carried out in order to be able to carry out synthetic broadband excitation by means of parameters of a narrowband codec.
• Another way to generate an excitation signal is to modulate the narrowband signal s nb (k) with a sine function at a fixed frequency or by directly using an arbitrary signal n eb (k), as already defined above was, is performed. It should be emphasized that the method used for the generation of the excitation signal is completely independent of the generation of the digital signal BWE and the format of this digital signal BWE and the decoding of this digital signal BWE. Therefore, can In this regard, an independent setting be performed.
• the digital signal BWE is decoded in the block 52 and the parameters of the signal power characterizing the temporal envelope and the spectral envelope, which are calculated according to the formulas 2) and 3), corresponding to the signals j (v ) and s p ( ⁇ > ⁇ ) .
• a reconstructive shaping of the temporal envelopes is first carried out in the exemplary embodiment. This is done in the decoder area 53. For this purpose, the excitation signal S exc ik) and the signal j (v) are transmitted to this decoder area 53. As shown in FIG.
• the excitation signal s exc [k ] is transmitted both to a block 531 and to a multiplier 532.
• the signal -J (v) is also transmitted to the block 531. From these signals transmitted to block 531, a scalar correction factor gi (k) is generated.
• This scalar correction factor gi (k) is transferred from the block 531 to the multiplier 532.
• the excitation signal s exc [k ] then becomes scalar
• Correction factor gi (k) multiplied and generates an output signal S 0x [Ic], which characterizes the reconstructed shaping of the temporal envelope.
• This output signal s exc [k] has the approximately correct temporal envelope, but is still inaccurate or imprecise with respect to the correct frequency, which in a subsequent step, the performing a reconstructed shaping of the spectral envelope is required to this imprecise frequency to be able to adjust the required frequency.
• the output signal S 0x [Ic) is transmitted to a second decoder area 54 of the decoder 5, to which the signal ⁇ ( ⁇ ⁇ ) is also transmitted.
• the second decoder area 54 has a block 541 and a Block 542, wherein the block 541 is designed to filter the output signal S 0x (Ic). From the output signal s exc (k) and the signal ⁇ ( ⁇ ⁇ ) , an impulse response h (k) is generated, which is transmitted from block 541 to block 542.
• the reconstructive shaping of the spectral envelope is then carried out from the output signal s exc (k) and d of the impulse response h (k). This reconstructed spectral envelope is then characterized by the output s exc (k) of block 542.
• a reconstructing shaping of the temporal envelope in a third decoder area 55 of the decoder 5 is again carried out.
• This reconstructing shaping of the temporal envelope takes place analogously as it is carried out in the first decoder region 53.
• a second scalar correction factor g 2 (k) is generated by the block 551 from the output signal s exc (k) and the signal J (v) , which is transmitted to a multiplier 552.
• the signal s eb (k) characterizing the signal components required for the bandwidth extension is then provided.
• This signal s eb (k) is transmitted to a summer 56, to which also the narrowband signal s nb (k) is transmitted.
• the bandwidth-extended output signal s w ° b (k) is generated and provided as the output signal of the decoder 5.
• the embodiment shown in FIG. 2 is merely exemplary and that the invention already has a single reconstructive shaping of the temporal envelopes, as is done in the first decoder region 53, and a single reconstructive shaping of the spectral envelopes, as in the second decoder region 54 carried out is sufficient. It should also be noted that it can also be provided that the reconstructive shaping of the spectral envelope in the second decoder area 54 is performed before the reconstruction of the temporal envelope in the first decoder area 53. This means that the second decoder region 54 is arranged before the first decoder region 53 in such an embodiment.
• the invention is advantageously used in the exemplary embodiment for a wideband input speech signal having a frequency range of about 50 Hz to 7 kHz.
• the invention is provided in the exemplary embodiment for the artificial extension of the bandwidth of speech signals, wherein the extension band is predetermined by the frequency range of about 3.4 kHz to about 7 kHz.
• the invention is used for an extension band, which is located in a low-frequency frequency range.
• the extension band may comprise a frequency range of about 50 Hz or even lower frequencies, up to a frequency range of about 3.4 kHz.
• the method according to the invention for the artificial extension of the bandwidth of speech signals can also be used such that the extension band comprises a frequency range which is at least partially above a frequency of about 7 kHz and for example up to 8 kHz, in particular 10 kHz , or even higher.
• the extension band comprises a frequency range which is at least partially above a frequency of about 7 kHz and for example up to 8 kHz, in particular 10 kHz , or even higher.
• a reconstructed formation of the temporal envelope in the first decoder area 53 is riert according to FIG 2 by a multiplication of the scalar first correction factor gi (k) and the excitation signal S ⁇ x (Jc) generation. It should be noted that a multiplication in
• the first scalar correction factor or gain gi (k) should have strict low-pass frequency characteristics.
• the excitation signal S ⁇ 0 (It) is segmented and analyzed in a manner already described above for the segmentation and the analysis of the extraction of the temporal envelope or the generation of the Signal s pfy) from the signal s eb (k) is performed in the encoder 1 by means of the block 12.
• the ratio between the decoded signal power as calculated by formula 3) and the analyzed result of the signal strength P ⁇ fy) results in a desired gain ⁇ (v) for the vth signal segment.
• This amplification factor of the vth signal segment is calculated in accordance with the following formula 6):
• the amplification factor or first correction factor gi (k) is calculated by interpolation and low-pass filtering. Low-pass filtering is crucial to This gain factor or this first correction factor gi (k) to limit the spectral envelope.
• the reconstructive shaping of the spectral envelope of the required signal components of the extension band is determined by filtering the output signal S 0x (Ic), which characterizes the reconstructed shaping of the temporal envelope.
• the filter operation can be implemented in the period or in the frequency domain.
• the corresponding frequency characteristic H (z) can be smoothed.
• the output signal s exc (k) of the first decoder region 53 is analyzed in order to be able to find the signal powers of the Pf ° ( ⁇ i, X).
• the desired amplification factor ⁇ ( ⁇ , ⁇ ) of a corresponding subband of the frequency range of the expansion band is calculated according to the following formula 7):
• the frequency characteristic H ( ⁇ , i) of the shape filters of the spectral envelope can be calculated by interpolation of the amplification factor ⁇ ( ⁇ , ⁇ ) and with a smoothing taking into account the frequency. If the shaping filter of the spectral envelope is to be used in the period, for example by a linear phase FIR filter, the filter coefficients can be calculated by an inverse FF transformation of the frequency characteristic H ( ⁇ , i) and a subsequent windowing.
• the reconstructive shaping of the temporal envelope influences the reconstructive shaping of the spectral envelopes and vice versa. It is therefore advantageous that, as explained in the exemplary embodiment and shown in FIG. provides an alternate performance of reconstructing a temporal envelope and a spectral envelope in an iterative process. Thereby, a substantially improved match of the temporal and spectral envelopes of the signal components of the enhancement band, which are reconstructed in the decoder and the corresponding temporal and spectral envelopes generated in the coder, can be achieved.
• one and a half times the iteration (reconstruction of the temporal envelopes, reconstruction of the spectral envelopes and renewed reconstruction of the temporal envelopes) is carried out.
• Bandwidth expansion facilitates the generation of an excitation signal having harmonics at the correct frequency, for example at an integer multiple of the fundamental frequency of the current sound.
• the invention can also be applied to downsampled subband signal components of the broadband input signal. This is advantageous when a low computational effort is required.
• the encoder 1 and the blocks 2 and 3 are arranged in a transmitter, wherein logically, the process steps carried out in the blocks 2 and 3 and the encoder 1 are then also carried out in the transmitter.
• the block 4 as well as the decoder 5 can advantageously be arranged in a receiver, whereby it is also clear that the preliminary steps carried out in the decoder 5 and in the block 4 are executed in the receiver.
• the invention can also be implemented in such a way that the method steps carried out in the coder 1 are carried out in the decoder 5 and are thus carried out exclusively in the receiver.
• the signal powers, which are calculated according to the formulas 2) and 3), in the deco 5 can be estimated.
• the block 52 is designed to estimate these parameters of the signal powers. This embodiment allows the concealment of potential transmission errors of the side information transmitted in the digital signal BWE.
• Estimation of lost parameters of an envelope for example by a loss of data, can be a troublesome switching of the signal bandwidth can be prevented.
• the invention In contrast to the known methods for artificially widening the bandwidth of speech signals, in the invention no transfer of already used amplification factors and filter coefficients is carried out as secondary information, but only the desired temporal and spectral envelopes are transmitted as side information to a decoder. Gain factors and filter coefficients are only then calculated in the decoder, which is arranged in a receiver. It can thereby be achieved that the artificial extension of the bandwidth in the receiver can be analyzed and, if necessary, corrected in a low-effort manner.
• the method according to the invention and the device according to the invention are very robust against disturbances of the excitation signal, whereby, for example, such a disturbance of a received narrowband signal can be caused by transmission errors.
• the transmission and the reconstructing shaping of the temporal and spectral envelopes separately, a very good resolution or splitting in the time domain and in the frequency domain can be achieved both in the time domain and in the frequency domain. This leads to a very good reproducibility of both stationary sounds and sounds as well as transient or short-term signals.
• the reproduction of stop consonants and plosives benefits from the significantly improved time resolution.
• the invention allows the frequency shaping to be performed by linear phase FIR filters rather than LPC synthesis filters.
• the invention allows a very flexible and modular design, which also allows the individual blocks in the receiver or in the Decoder 5 can advantageously be exchanged or set in an advantageous manner For such a change or adjustment, no change of the transmitter or the coder 1 or the format of the transmission signal with which the coded information to the decoder 5 or the receiver ü
• different decoders can be operated with the method according to the invention, as a result of which a restoration of the broadband input signal can be carried out with different precision as a function of the available computing power.
• the received parameters which characterize the spectral and temporal envelopes can be used not only for an extension of the bandwidth, but also for the support of subsequent signal processing blocks, such as post-filtering, or additional coding steps such as Transformer encoder, can be used.
• the resulting narrowband speech signal s nb (k), as available to the bandwidth expansion algorithm, may be present, for example, after a reduction of the sampling frequency by a factor of 2 at a sampling rate of 8 kHz.
• the invention and the underlying principle of bandwidth expansion it is possible to generate a broadband excitation of information of the G.729AH standard.
• the data rate of the secondary signals transmitted in the digital signal BWE Information can be about 2 kbit / s.
• a relatively low-complexity calculation system or a relatively low complex computational effort is required, which is less than 3 WMOPS.
• the inventive method and the device according to the invention is very robust against baseband disturbances of the G.729AH standard.
• the invention may also be used advantageously for use in voice-over-IP.
• the method according to the invention and the device according to the invention are compatible with TDAC envelopes.
• the invention also has a very modular and flexible structure and a modular and flexible conception.

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  • 2006-06-30 DK DK06840370T patent/DK1825461T3/da active
  • 2006-06-30 AT AT06840370T patent/ATE407424T1/de not_active IP Right Cessation

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