Patent Application
20130028409
The present invention relates to the field of call voice signal processing, and particularly, to an apparatus and a method for echo control in a parameter domain.
The call voice quality is one of the critical indexes for measuring the telecom network, and also a core competence of the telecom equipment manufacturers. In order to ensure the voice quality during the call, a key point is to effectively process the echoes and other interference sources generated during the call, so as to improve the subjective feelings of both parties concerned in the call.
The echoes generated during the call may be classified into electric echo and acoustic echo according to their sources. In which, the mechanism for producing the electric echo is illustrated in
The conventional echo control algorithm is implemented based on the Pulse Code Modulation (PCM) in the linear domain. The so called linear domain is a mode based on direct sample point value, and when the communication link has a codec compression part, corresponding decoder is required to completely decode the compressed code stream and recover to the sample point value. It is adaptive to the scenario where the passed network elements require a decoding or the input is PCM in the linear domain. However, with the occurrences of the Tandem Free Operation (TFO) and the Transcoder Free Operation (TrFO) in recent years, it is not required to perform multiple times of transforms of speech codec between the transmitting and receiving ends, thus the conventional echo control algorithm is no longer applicable, and characteristic parameters in the parameter domain shall be obtained through decoding in the parameter domain to perform an echo control in the parameter domain. The so called decoding in the parameter domain means only partially decoding the compressed code stream to extract the characteristic parameters in the parameter domain of each frame, including fixed codebook gain, adaptive codebook gain, line spectrum frequency, etc. The echo control is implemented by modifying the characteristic parameters in the parameter domain in the compressed code stream, without recovering the compressed code stream to the sample point value through the decoder.
When an echo control is performed in the parameter domain, the input and output are all compressed code signals, thus for some encoding types such as Adaptive Multi Rate Codec (AMR) and Enhanced Full Rate Codec (EFR), the frames are associated with each other, and it shall not simply and independently perform the echo control for a certain frame. Particularly, during a handover between echo and non-echo, some special processing is required to achieve a more natural transition between echo and non-echo.
The prior art provides a method for echo control applicable to TFO/TrFO scenarios. The method performs a transition between echo and non-echo by re-quantizing and outputting the fixed codebook gain and the adaptive codebook gain in the near-end input signal.
However, during the transition between echo and non-echo in the prior art, when related processing of the code type signals such as AMR and EFR are to be made, the quantization of a parameter is associated with the prediction error of the parameter in the previous frame because the frames of those signals are associated with each other, while the prior art does not re-quantize the related linear prediction coefficients. Thus when the signals using the above coding mode are processed in the prior art, there is a risk that the finally decoded linear prediction parameters may be abnormal, resulting in signal mutations, then the handover between echo and non-echo cannot be smoothly transited.
An objective of the present invention is to provide an apparatus for echo control in a parameter domain, and intended to solve the problem in the prior art that the handover between echo and non-echo cannot be transited naturally when the transition from echo and non-echo is treated.
The embodiments of the present invention are implemented as follows. An apparatus for echo control in a parameter domain, comprising:
an echo detection module configured to extract parameter domain characteristic parameters of a far-end output signal and a near-end input signal, respectively, through a parameter domain decoding, and detect whether a near-end input signal frame is an echo frame according to the parameter domain characteristic parameters, wherein the parameter domain characteristic parameters comprise a Line Spectrum Frequency (LSF), a fixed codebook gain and an adaptive codebook gain; and
a transition module configured to re-quantize the fixed codebook gain, the adaptive codebook gain and an LSF prediction error in the near-end input signal frame when the echo detection module detects the near-end input signal frame as a non-echo frame, and replace the original fixed codebook gain, the original adaptive codebook gain and the original LSF in the near-end input signal frame.
Another objective of the present invention is to provide a method for echo control in a parameter domain, comprising:
detecting whether a near-end input signal frame is an echo frame according to parameter domain characteristic parameters of a far-end output signal and a near-end input signal, wherein the parameter domain characteristic parameters comprise LSF, fixed codebook gain and adaptive codebook gain; and
when the near-end input signal frame is detected as an echo frame, re-quantizing the fixed codebook gain, the adaptive codebook gain and the LSF prediction error in the near-end input signal frame to replace the original fixed codebook gain, the original adaptive codebook gain and the original LSF in the near-end input signal frame, respectively.
The embodiments of the present invention control the echo generated during the call in the parameter domain. As compared with the conventional method for echo control in the linear domain which requires the compressed code stream to be completely decoded, the present invention greatly improves the echo control efficiency. When a non-echo frame is detected, the handover between echo and non-echo is performed by re-quantizing the fixed codebook gain and the adaptive codebook gain in the non-echo frame. Meanwhile, considering that the LSF in the non-echo frame is associated with the LSF prediction error of the previous frame, the LSF prediction error of the non-echo frame is re-quantized to replace the original LSF, which reduces the risk that the LSF finally obtained by decoding may be abnormal, thereby avoiding the signal mutation, and realizing a smooth transition during the handover between echo and non-echo.
In the parameter domain, during the control of echo generated in a call, when a handover between echo and non-echo is to be made, according to the embodiment of the present invention, the LSF prediction error of the current frame is re-quantized and output as the LSF in the near-end output signal, thus the association between the frames is taken into account during the handover process, thereby avoiding a hard handover between echo and non-echo caused by independently processing the current frame, and improving the transition effect between echo and non-echo.
Referring to
The echo detection module 41 extracts parameter domain characteristic parameters of a far-end output signal and a near-end input signal, respectively, through a parameter domain decoding, and detects whether the near-end input signal is an echo according to the extracted parameter domain characteristic parameters.
The partial decoding unit 411 is configured to perform a partial decoding for the inputted far-end output signal and near-end input signal, respectively, so as to extract parameter domain characteristic parameters of the current sub-frames of the far-end output signal and the near-end input signal, including LSF, pitch period, fixed codebook gain, adaptive codebook gain, energy and other parameters.
The cross-correlation unit 412 is configured to take the two groups of parameter domain characteristic parameters of the near-end input signal and the far-end output signal extracted by the partial decoding unit 411 as near-end vectors and far-end (interval [fixed delay, fixed delay+dynamic delay]) vectors, respectively, perform a cross-correlation operation on the two groups of vectors to obtain a series of cross-correlation coefficients, and record the timing corresponding to each group of cross-correlation coefficients.
The sub-frame judgment unit 413 is configured to record the timing in a far-end interval corresponding to the maximum value of the series of cross-correlation coefficients as an initial echo delay, and judge whether both the far-end pitch period and near-end pitch period at this timing are less than a first predetermined threshold (i.e., judging the similarity between the far-end and near-end pitch periods), or judge whether both the far-end and near-end energies at this timing are less than a second predetermined threshold; and if yes, the sub-frame judgment unit 413 determines the current sub-frame of the near-end input signal as an echo sub-frame, otherwise determines it as a non-echo sub-frame.
To be noted, the partial decoding unit 411, the cross-correlation unit 412 and the sub-frame judgment unit 413 all make the calculations and judgments based on each sub-frame in the current frame of the far-end output signal and the near-end input signal.
The echo judgment unit 414 is configured to perform a synthetic judgment based on the judgment results of each sub-frame in the current frames, to judge whether the current frame of the near-end input signal is an echo frame or a non-echo frame, after the partial decoding unit 411, the cross-correlation unit 412 and the sub-frame judgment unit 413 all make the calculations and judgments for all sub-frames of the current frame.
In this embodiment, the synthetic judgment made by the echo judgment unit 414 shall firstly judge whether the current near-end input frame is a Silence Insertion Descriptor (SID) frame; if yes, determine the current near-end input frame as a non-echo frame regardless of whether any echo sub-frame is included; and if the current near-end input frame is not a SID frame, it is to perform the judgement based on a “majority rule”. When the number of echo sub frames in the near-end input frame is larger than that of non-echo sub frames, it is determined the current near-end input frame is an echo frame. On the contrary, it is determined the current near-end input frame is a non-echo frame when the number of non-echo sub frames in the near-end input frame is larger than that of echo sub frames. The process of judging whether the current frame is an SID frame is performed by a near-end input judgment unit 415 to be described later, and herein is omitted.
According to an embodiment of the present invention, as illustrated in
The near-end input judgment unit 415 is configured to perform an independent judgment on the near-end input signal, so as to judge whether the current frame of the near-end input is a silence frame by detecting whether the flag of the current frame in the near-end input signal indicates the current frame as an SID frame. In case the current frame is not judged as a silence frame, it is synthetically judged whether the current frame is a background noise frame based on the parameters obtained by partially decoding the near-end input signal, such as the calculated signal to noise ratio, frequency spectrum, energy, etc.
The cache unit 416 is configured to cache LSFs obtained by a partial decoding when the current frame of the near-end input signal is a silence frame or a background noise frame. In the embodiment, the cache unit 416 circularly caches the inputted LSFs, i.e., it only reserves eight frames of LSFs nearest to the current time.
The energy accumulation unit 417 is configured to perform a long-term average accumulation of the current frame energy obtained by partially decoding the near-end input signal, when the current frame of the near-end input signal is a silence frame or a background noise frame.
The results output by the cache unit 416 and the energy accumulation unit 417 are input into the comfort noise insertion module 42, so that the echo frame can be replaced more gently and naturally. The implementation will be described later, and herein is omitted.
In this embodiment, after the echo detection module 41 detects whether the near-end input signal is an echo, a switch A in
The echo frame may be cancelled in a comfort noise insertion method or an echo attenuation method, herein the comfort noise insertion method is taken as an example. Generally during the call, about 40% of the time is the silent time, i.e., in about 40% of the time only the ambient noise rather than any voice information is transmitted, thus most part of those signals are mainly useless signals. If those signals are encoded in another encoding mode of a lower bit rate, the overhead of controlling the echo can be reduced and the bandwidth can be saved. Currently, the encoding modes such as AMR and EFR all introduce the discontinuous emission mechanism.
Based on the above principle, the comfort noise insertion module 42 replaces the frame determined as echo in the echo detection module 41 with a silence frame or a background noise frame, encodes it using an SID encoding mode with a low bit rate, and then outputs the encoded frame.
Referring to
The LSF generation unit 421 is configured to generate a prior-quantization LSF to replace an LSF of the echo frame. During the generation, the LSF generation unit 421 firstly judges the current cache condition of the cache unit 416 to determine whether eight frames of LSFs are cached therein; if yes, the LSF generation unit 4 randomly selects one frame of LSF from the cache unit 416 and takes it as the prior-quantization LSF to replace the LSF of the echo frame. If the cached LSFs are less than eight frames, the LSF generation unit 4 transforms each frame of LSF having been cached into a Line Spectral Pair (LSP) and accumulates them to obtain an LSP average value, then transforms the LSP average value into an LSF which acts as the Prior-quantization LSF to replace the LSF of the echo frame.
The LSF quantization unit 422 is configured to perform corresponding quantization of the LSF outputted by the LSF generation unit 421 according to different encoding types, and then replace the original LSF of the near-end input signal.
Through the above processing, the echo frame is replaced with an SID frame with a low bit rate, thus the echo control overhead is reduced and the bandwidth is saved. In addition, the time domain optimization effect achieved by replacing the echo frame with the SID frame is illustrated in
According to an embodiment of the present invention, the comfort noise insertion module 42 further includes an energy quantization unit 423.
The energy quantization unit 423 is configured to perform corresponding quantization of the average value of the long-term accumulated energy in the energy accumulation unit 417 according to the different encoding types, so as to replace the energy part of the near-end output signal (for the EFR encoding type, the energy part is corresponding to the fixed codebook gain part of the near-end output signal), thus the frame energy after the echo replacement is changed slowly and transited naturally, and the subjective experiences of the parties concerned in the call can be greatly improved.
In this embodiment, when the echo detection module 41 judges the current frame of the near-end input signal as a non-echo frame, the output of the echo detection module 41 is inputted into the transition module 43 in which the prediction error of the LSF of the current frame in the near-end input signal is re-quantized, thus the relationships between the frames are sufficiently considered, which greatly decreases the risk that there may be abnormal during the transition from echo to non-echo in the encoding modes such as AMR and EFR. The module is a key point of the present invention, and its specific structure is illustrated in
Referring to
The LSF prediction error calculation unit 431 firstly multiplies the LSF prediction error of a previous frame in the near-end input signal with an LSF prediction error coefficient, and adds the multiplication result with an LSF average value to obtain an LSF prediction value of the current frame. Next, the LSF prediction error calculation unit 431 subtracts the LSF prediction value from the LSF obtained by partially decoding the near-end input signal to acquire an LSF prediction error of the current frame prior to quantization, and caches the LSF prediction error to be used in the LSF calculation for a next frame. In the embodiment, the LSF prediction error coefficient and the LSF average value are determined according to different signal encoding modes, and they are corresponding constants for different encoding modes.
In the embodiment of the present invention, through the LSF prediction error calculation unit 431, the relationships between the frames are sufficiently considered, which greatly decreases the risk that there may be abnormal in the transition from echo to non-echo under the encoding modes such as AMR and EFR. To be noted, the above LSF calculation mode is not limited to the encoding mode AMR or EFR, and it is also adaptive to the handover between echo and non-echo during an echo control in the parameter domain under other encoding mode.
The codebook gain calculation unit 432 is configured to limit the fixed codebook gain and the adaptive codebook gain obtained by partially decoding the near-end input signal. In order to avoid the abnormity caused by a waveform mutation during the handover between echo and non-echo, if the fixed codebook gain or the adaptive codebook gain obtained by the partial decoding exceeds 1, it shall be set as 1, so as to ensure that the excitation after the near-end output signal is decoded will not be mutated during the transition from echo to non-echo.
The quantization unit 433 is configured to re-quantize the LSFs and the codebook gains outputted by the LSF prediction error calculation unit 431 and the codebook gain calculation unit 432, respectively, and replace the original LSF, fixed codebook gain and adaptive codebook gain of the near-end input signal frame, respectively.
To be particularly pointed out, in the embodiment when a transition is to be made, the transition module 43 only modifies the LSF, the fixed codebook gain and the adaptive codebook gain in the near-end input signal, and then re-quantizes and outputs them, while other parameters will not be changed.
As an embodiment of the present invention, referring to
The transition selection unit 434 is configured to determine whether a non-echo frame currently inputted into the transition module 43 needs to be transited.
The parameter recording unit 435 is configured to record some parameter domain characteristic parameters (e.g., the LSF prediction error to be used by the LSP calculation unit 431) obtained by partially decoding the non-echo frame, when the transition selection unit 434 determines that the non-echo frame currently inputted into the transition module 43 does not need to be transited, i.e., the previous frame of the current frame of the near-end input signal is also a non-echo frame. After that, the non-echo frame is outputted through a direct transparent transmission, and the LSF, the fixed codebook gain and the adaptive codebook gain no longer need to be recalculated, thus the processing overhead for echo control is well decreased.
The state judgment unit 436 is configured to perform an inverse quantization of the near-end input signal outputted by the quantization unit 433, and compares the result with the original near-end input signal to judge whether their LSFs and codebook gains are completely consistent with each other; if yes, the comparison result is outputted to the state judgment unit 436 so that the state judgment unit 436 selects the internal state of the transition module 43 as not requiring a transition, otherwise the current state of the transition module 43 is kept.
Meanwhile, in the embodiment, when the internal state of the transition module 43 becomes a state of not requiring a transition, the transition selection unit 434 judges the result outputted by the echo detection module 41. Once the echo detection module 41 outputs an echo frame, the transition selection unit 434 again make the transition module 43 enter the internal state of requiring a transition, so as to implement a handover from echo to non-echo in time when the next non-echo frame arrives.
Thus in the embodiment of the present invention, when the echo of the near-end input signal does not occur for a long time, the non-echo frames are directly transmitted transparently through a transition selection, and it is unnecessary to calculate the LSF, the fixed codebook gain and the adaptive codebook gain for each frame of the near-end input signal. Further, when a handover is to be made between echo and non-echo, a natural transition is well realized, which greatly decreases of the disk of abnormity occurring, and improves the subjective experiences of the parties concerned in the call.
Finally, the near-end input signal outputted by the comfort noise insertion module 42 or the transition module 43 is inputted into the bit packing module 44, which performs bit packing for the inputted signal according to the information such as frame type and frame rate. Since the bit packing is performed according to the standard protocol, and there are relatively fixed procedures, and it is omitted.
The units included in the above apparatus embodiment are just classified based on the functional logics, but not limited thereto so long as corresponding functions can be realized. In addition, the specific titles of the functional units are also given for the convenience of distinguishing them from each other, rather than limitations to the protection scope of the present invention.
In the embodiment of the present invention, the above described apparatus for echo control in the parameter domain may construct an echo control system of a wireless network together with a wireless base station, a base station controller, a Universal Media Gateway (UMG) and other network elements of gateway type, or construct an echo control system of a PSTN network together with devices such as the Public Switched Telephone Network (PSTN) switchboard. The apparatus provides uplink and downlink communication interfaces, through which uplink and downlink echo control functions in the parameter domain can be provided.
According to an embodiment of the present invention, as illustrated in
According to an embodiment of the present invention, as illustrated in
Step S1201: detecting whether a near-end input signal frame is an echo frame according to parameter domain characteristic parameters of a far-end output signal and a near-end input signal, wherein the parameter domain characteristic parameters include LSF, pitch period, fixed codebook gain, adaptive codebook gain and energy.
The embodiment of the present invention performs the echo control in the parameter domain, and the echo can be controlled just by modifying the parameter domain characteristic parameters in the signals, thus firstly the parameter domain characteristic parameters for the echo control are extracted from the far-end output signal and the near-end input signal.
Step S1202: when the near-end input signal frame is detected as an echo frame, replacing the near-end input signal frame with a silence frame or a background noise frame.
The silence frame and the background noise frame are both transmitted in the SID encoding mode with a low bit rate. Thus in the embodiment of the present invention, when an echo frame occurs and an echo cancellation is required, the echo frame is replaced with the silence frame or the background noise frame, thereby reducing the echo control overhead and saving the bandwidth.
Step S1203: when the near-end input signal frame is detected as a non-echo frame, re-quantizing the fixed codebook gain, the adaptive codebook gain and an LSF prediction error in the near-end input signal frame, so as to replace the original fixed codebook gain, adaptive codebook gain and LSF in the near-end input signal frame.
Step S1204: bit-packing and outputting the modified near-end input signal according to the frame type and the frame rate of the near-end input signal frame.
In the embodiment of the present invention, through the above steps, when a handover between echo and non-echo is required during the echo control in the parameter domain, since the LSF prediction error of the currently controlled input signal frame is re-quantized and outputted as an LSF in the near-end output signal, thus the association between the LSF quantization in the current frame and the LSF prediction error in the previous frame is considered during the handover process, thereby avoiding a hard handover between echo and non-echo caused by independently processing the current frame, and improving the transition effect between echo and non-echo.
Next, the flows for implementing the method for echo control in the parameter domain provided by the embodiment of the present invention are specifically described as follows one by one.
Step S1301: extracting parameter domain characteristic parameters of the current sub-frames of the near-end input signal and the far-end output signal, respectively, and taking them as near-end vectors and far-end (interval [fixed delay, fixed delay+dynamic delay]) vectors, respectively.
Step S1302: performing a cross-correlation operation on the near-end vectors and far-end vectors to obtain groups of cross-correlation coefficients, and recording the timing corresponding to each group of cross-correlation coefficients.
Step S1303: taking the timing corresponding to the maximum cross-correlation coefficient as an initial echo delay; thus, judging whether the current sub-frame is an echo sub-frame by detecting the similarity between the far-end and near-end pitch periods at this timing, or by detecting whether the difference between the far-end and near-end energies is below a predetermined threshold; and if yes, the current sub-frame of the near-end input signal is an echo sub-frame, otherwise it is a non-echo sub-frame.
The above steps S1301 to S1303 are all processing of the sub-frame, thus in step S1304, a synthetic judgment is made based on the echo detection result of each sub-frame in the near-end input signal frame, so as to judge whether the near-end input signal frame is an echo frame.
Step S1401: judging whether the near-end input signal frame is a silence frame or a background noise frame, and if yes, caching eight frames of LSFs in the silence frame or the background noise frame of the near-end input signal nearest to the current timing.
Step S1402: judging whether the cached LSFs reach eight frames, and if yes, randomly selecting one frame of LSF to generate a prior-quantization LSF, otherwise transforming each cached frame of LSF into a Line Spectral Pair (LSP) and accumulating to obtain an LSP average value, then transforming the LSP average value into an LSF to generate the prior-quantization LSF.
Step S1403: quantizing the generated prior-quantization LSF to replace the original LSF of the near-end input signal frame.
Meanwhile, in the embodiment when the near-end input signal frame is the silence frame or the background noise frame, a long-term average accumulation of the energy of the silence frame or the background noise frame of the near-end input signal is carried out to generate an energy long-term accumulation average value, which is quantized to replace the original energy of the near-end input signal frame, so that the frame energy after the echo replacement is changed slowly and transited naturally.
Step S1501: judging whether the fixed codebook gain or the adaptive codebook gain of the non-echo near-end input signal frame exceeds 1, and if yes, limiting its value as 1.
Through the above limitation, it is ensured that the excitation after the near-end output signal is decoded will not be mutated during the transition from echo to non-echo.
Step S1502: multiplying the LSF prediction error of a previous near-end input signal frame with an LSF prediction error coefficient, and adding the multiplication result with an LSF average value to obtain an LSF prediction value of the current near-end input signal frame; subtracting the LSF prediction value from the LSF of the near-end input signal frame to obtain an LSF prediction error of the current near-end input signal frame prior to quantization, wherein the LSF prediction error coefficient and the LSF average value are determined by the encoding mode of the near-end input signal frame.
Step S1503: re-quantizing the codebook gain and the LSF prediction error obtained in the above step.
Further, after the above steps are ended, as illustrated in
During a call echo control in the parameter domain by the embodiment of the present invention, when a handover between echo and non-echo is required, it is realized by re-quantizing the fixed codebook gain and the adaptive codebook gain of the current frame. Meanwhile, the LSF prediction error of the current frame is re-quantized to replace the original LSF output in the near-end output signal. Thus the association between the frames is considered during the handover, thereby avoiding a hard handover between echo and non-echo caused by independently processing the current frame, and improving the transition effect between echo and non-echo.
A person skilled in the art shall be appreciated that in the method for echo control in the parameter domain provided by the embodiment of the present invention, all or a part of the steps may be completed by instructing relevant hardware through a program, for example by running a program in the computer. The program may be stored in a readable access medium, such as RAM, magnetic disk, optical disk, etc.
This application is a continuation of International Application No. PCT/CN2011/077587, filed on Jul. 25, 2011, which is hereby incorporated by reference in its entirety.
| Number | Date | Country | |
|---|---|---|---|
| Parent | PCT/CN2011/077587 | Jul 2011 | US |
| Child | 13555502 | US |
