CN100421438C - Bit loading method in selecting frequency single carrier wave blocking transmission system - Google Patents

Bit loading method in selecting frequency single carrier wave blocking transmission system Download PDF

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CN100421438C
CN100421438C CNB2005100420564A CN200510042056A CN100421438C CN 100421438 C CN100421438 C CN 100421438C CN B2005100420564 A CNB2005100420564 A CN B2005100420564A CN 200510042056 A CN200510042056 A CN 200510042056A CN 100421438 C CN100421438 C CN 100421438C
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snr
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CN1649333A (en
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杜岩
袁静
李剑飞
宫良
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Shandong University
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Abstract

The present invention provides a bit loading method for a frequency selection single carrier blocking transmission system. The present invention comprises steps that (1) communication is established between a receiving party and a sending party, a receiving terminal obtains noise power after equalization according to the number M of selected available sub-channels and the amplitude gain of the channels, and the signal-to-noise ratio of a system after equalization is obtained according to a receiving signal-to-noise ratio and an equalization mode; (2) different linear modulation modes used by the system are determined according to the signal-to-noise ratio of the system after equalization, channel estimation errors and signal-to-noise ratio loss caused by synchronization errors, data which describes the different linear modulation modes is call as the information of modulation modes, and the information of modulation modes is sent to a sending terminal through a feedback channel; (3) the sending terminal sends signals through symbol mapping according to the received information of modulation modes; (4) the receiving terminal demodulates the signals according to the information of modulation modes and makes a judgment. The frequency spectrum efficiency and the power efficiency of the present invention are superior to those of the existing SC-FDE and OFDM systems under the condition of ensuring the performance stabilization of the system.

Description

Bit loading method in frequency-selecting single carrier block transmission system
(I) technical field
The invention relates to a broadband digital communication transmission method. Belongs to the technical field of broadband wireless communication.
(II) background of the invention
Communication technology has been developed over the last decades, particularly the nineties of the twentieth century, with profound effects on the development of people's daily lives and national economy. In the future, communication technologies are developing towards high-speed broadband, so that many broadband digital transmission technologies are receiving wide attention, and Orthogonal Frequency Division Multiplexing (OFDM) and single carrier with Frequency Domain Equalization (SC-FDE) are two broadband digital transmission technologies that are regarded by people, and both of them belong to block transmission technologies, while OFDM is far more concerned than SC-FDE at present, and become support technologies in various standards, for example: IEEE802.11a in a Wireless Local Area Network (WLAN); IEEE802.16 in Wireless Metropolitan Area Network (WMAN: Wireless Metropolisan Area Network); various high-speed Digital Subscriber lines (xDSL) in wired data transmission are standards based on OFDM technology. SC-FDE is not adopted by these standards, but is proposed as a physical layer transmission technique in IEEE802.16 in combination with OFDM.
An OFDM system is a multi-carrier transmission technique that uses N subcarriers to divide the entire wideband channel into N parallel mutually orthogonal narrowband subchannels. OFDM systems have a number of compelling advantages: 1. very high spectral efficiency; 2. the realization is simpler; 3. the anti-multipath interference capability and the anti-fading capability are strong; 4. channel state information (i.e., adaptive OFDM techniques) may be utilized to further improve spectral efficiency, etc.
The adaptive OFDM technology can adjust the bit number distributed on different frequency domain points (namely different sub-channels) according to the given input signal power and the channel condition, control the bit error rate on each sub-channel to be basically the same, enable the bit error rate to meet the system performance requirements, and improve the system transmission code rate as much as possible, thereby realizing the bit loading (bit-loading) of the OFDM system. The self-adaptive OFDM technology can fully utilize the channel state information and improve the spectrum efficiency of the system.
It is these advantages that make OFDM a hot research topic in the last decade and is considered as a supporting technology for future communications, especially broadband wireless communications. However, many disadvantages of the OFDM system itself, especially its Peak-to-Average Power Ratio (PAPR) is too large, which limits its practical pace, and the existing SC-FDE has all the advantages of the OFDM except the fourth one, and does not have the PAPR problem of the OFDM, and the performance and efficiency are basically equivalent to the OFDM. The SC-FDE system is developed on the basis of researching OFDM, and adopts block transmission as OFDM and a Cyclic Prefix (CP) mode, so that the linear convolution of signals and channel impulse response can be converted into Cyclic convolution, and the interframe interference caused by multipath is eliminated. Thus, the simple frequency domain equalization technology is adopted at the receiving end to eliminate the intersymbol interference, for example: zero-forcing equalization and minimum mean square error equalization.
Compared with OFDM, the SC-FDE system has no PAPR problem. The PAPR problem is a problem that the OFDM system itself is difficult to solve in a low-cost manner (spectrum efficiency and power efficiency). SC-FDE technology is therefore currently receiving increasing attention. The mathematical model of the conventional SC-FDE system is briefly described below.
In the SC-FDE system, a frame of time domain signal transmitted by a transmitting end is s (N), (N is 0, 1, …, N-1), an impulse response of a channel is h (N), (N is 0, 1, … L-1) through a multipath channel, interference of Additive White Gaussian Noise (AWGN) is received during signal transmission, the Noise is w (N), (N is 0, 1, …, N-1), and after a CP is removed, a received time domain signal r (N) is:
<math> <mrow> <mi>r</mi> <mrow> <mo>(</mo> <mi>n</mi> <mo>)</mo> </mrow> <mo>=</mo> <mi>s</mi> <mrow> <mo>(</mo> <mi>n</mi> <mo>)</mo> </mrow> <mo>&CircleTimes;</mo> <mi>h</mi> <mrow> <mo>(</mo> <mi>n</mi> <mo>)</mo> </mrow> <mo>+</mo> <mi>w</mi> <mrow> <mo>(</mo> <mi>n</mi> <mo>)</mo> </mrow> <mo>,</mo> <mrow> <mo>(</mo> <mi>n</mi> <mo>=</mo> <mn>0,1</mn> <mo>,</mo> <mo>.</mo> <mo>.</mo> <mo>.</mo> <mo>,</mo> <mi>N</mi> <mo>-</mo> <mn>1</mn> <mo>)</mo> </mrow> <mo>-</mo> <mo>-</mo> <mo>-</mo> <mrow> <mo>(</mo> <mn>1</mn> <mo>)</mo> </mrow> </mrow> </math>
wherein,representing a circular convolution operation.
At the receiving end, the signal is transformed to the frequency domain by Discrete Fourier Transform (DFT), and the obtained frequency domain signal is as follows according to the time domain convolution theorem of DFT:
R(k)=S(k)·H(k)+W(k),(k=0,1,…,N-1) (2)
where, r (k), s (k), h (k), w (k) are r (N), s (N), h (N), w (N) are frequency domain symbols of N-point DFT, and h (k), (k ═ 0, 1, …, N-1) are frequency domain responses of the channel. After zero-forcing equalization, the frequency domain signals are:
S ~ ( k ) = S ( k ) + W ( k ) H ( k ) = S ( k ) + W ~ ( k ) , ( k = 0,1 , . . . , N - 1 ) - - - ( 3 )
and finally, performing Inverse Discrete Fourier Transform (IDFT) on the signal, converting the signal back to a time domain for judgment, and obtaining the data transmitted by the transmitting end.
As can be seen from equation (3), the finally obtained signal has an error with the transmitted real signal, and the error is caused by noise, and especially, the noise is excessively amplified in the case of a deep fading point of a channel, and the signal is distorted when the signal is equalized by the minimum mean square error. These problems can be alleviated if channel state information is utilized in the SC-FDE system. Therefore, the applicant proposes a frequency-selective single-carrier block transmission method (applied national invention patent, patent application number: 200410036439.6), which overcomes the disadvantage that the traditional SC-FDE system cannot utilize the channel state information, and the new SC-FDE system has higher system performance and efficiency.
The method for realizing the single carrier block transmission in the frequency selection mode comprises the following steps:
first, finding out available sub-channel, and using channel as mark, then sending sub-channel mark information to sending end through reverse channel.
The receiving end selects M (M is less than or equal to N) usable sub-channels from N sub-channels according to the estimated channel state information H (k), (k is 0, 1, …, N-1) and according to the amplitude gain, and the labels of the M usable sub-channels are ki(i ═ 0, 1, …, M-1), and the remaining subchannels are disabled, and it is marked with 1-bit information, i.e. "0" or "1", whether each subchannel is an available subchannel or a disabled subchannel, which is the subchannel marking information required by the transmitting end, if the receiving end makes N-point DFT, i.e. there are N subchannels in total, the subchannel marking information fed back to the transmitting end has N bits in total, and then the N-bit information is sent back to the transmitting end through the reverse channel.
Second, the signal spectrum is changed according to the subchannel flag information
After receiving the subchannel flag information sent back by the receiving end, the transmitting end can use M available subchannels to transmit signals, so that for a frame of M SC-FDE symbols s (n), (n ═ 0, 1, …, M-1), M-point DFT is performed to transform to the frequency domain:
<math> <mrow> <mi>S</mi> <mrow> <mo>(</mo> <mi>i</mi> <mo>)</mo> </mrow> <mo>=</mo> <munderover> <mi>&Sigma;</mi> <mrow> <mi>n</mi> <mo>=</mo> <mn>0</mn> </mrow> <mrow> <mi>M</mi> <mo>-</mo> <mn>1</mn> </mrow> </munderover> <mi>s</mi> <mrow> <mo>(</mo> <mi>n</mi> <mo>)</mo> </mrow> <msup> <mi>e</mi> <mrow> <mo>-</mo> <mi>j</mi> <mfrac> <mrow> <mn>2</mn> <mi>&pi;</mi> </mrow> <mi>M</mi> </mfrac> <mi>ni</mi> </mrow> </msup> <mo>,</mo> <mrow> <mo>(</mo> <mi>i</mi> <mo>=</mo> <mn>0,1</mn> <mo>,</mo> <mo>.</mo> <mo>.</mo> <mo>.</mo> <mo>,</mo> <mi>M</mi> <mo>-</mo> <mn>1</mn> <mo>)</mo> </mrow> <mo>-</mo> <mo>-</mo> <mo>-</mo> <mrow> <mo>(</mo> <mn>4</mn> <mo>)</mo> </mrow> </mrow> </math>
obtaining the frequency domain signal of M points, using the k-th selectedi(i-0, 1, …, M-1) available subchannels H (k)i) (i ═ 0, 1, …, M-1) the ith frequency domain signal S (i), (i ═ 0, 1, …, M-1) is transmitted, i.e. the frequency domain signal to be transmitted is placed at the signal spectrum point corresponding to the available subchannel, and the signal spectrum point corresponding to the forbidden subchannel is set to zero, or some non-information data may be filled, so as to obtain a new frame of frequency domain signal S' (k), (k ═ 0, 1, …, N-1), the number of points is N:
<math> <mrow> <msup> <mi>S</mi> <mo>&prime;</mo> </msup> <mrow> <mo>(</mo> <mi>k</mi> <mo>)</mo> </mrow> <mo>=</mo> <mfenced open='{' close=''> <mtable> <mtr> <mtd> <mi>S</mi> <mrow> <mo>(</mo> <mi>i</mi> <mo>)</mo> </mrow> <mo>,</mo> </mtd> <mtd> <mi>k</mi> <mo>=</mo> <msub> <mi>k</mi> <mi>i</mi> </msub> </mtd> </mtr> <mtr> <mtd> <mn>0</mn> <mo>,</mo> </mtd> <mtd> <mi>k</mi> <mo>&NotEqual;</mo> <msub> <mi>k</mi> <mi>i</mi> </msub> </mtd> </mtr> </mtable> </mfenced> <mo>,</mo> <mrow> <mo>(</mo> <mi>k</mi> <mo>=</mo> <mn>0,1</mn> <mo>,</mo> <mo>.</mo> <mo>.</mo> <mo>.</mo> <mo>,</mo> <mi>N</mi> <mo>-</mo> <mn>1</mn> <mo>)</mo> </mrow> <mo>-</mo> <mo>-</mo> <mo>-</mo> <mrow> <mo>(</mo> <mn>5</mn> <mo>)</mo> </mrow> </mrow> </math>
then, for S' (k), (k ═ 0, 1, …, N-1), an N-point IDFT is made:
<math> <mrow> <msup> <mi>s</mi> <mo>&prime;</mo> </msup> <mrow> <mo>(</mo> <mi>n</mi> <mo>)</mo> </mrow> <mo>=</mo> <mfrac> <mn>1</mn> <mi>N</mi> </mfrac> <munderover> <mi>&Sigma;</mi> <mrow> <mi>k</mi> <mo>=</mo> <mn>0</mn> </mrow> <mrow> <mi>N</mi> <mo>-</mo> <mn>1</mn> </mrow> </munderover> <msup> <mi>S</mi> <mo>&prime;</mo> </msup> <mrow> <mo>(</mo> <mi>k</mi> <mo>)</mo> </mrow> <msup> <mi>e</mi> <mrow> <mi>j</mi> <mfrac> <mrow> <mn>2</mn> <mi>&pi;</mi> </mrow> <mi>N</mi> </mfrac> <mi>nk</mi> </mrow> </msup> <mo>,</mo> <mrow> <mo>(</mo> <mi>n</mi> <mo>=</mo> <mn>0,1</mn> <mo>,</mo> <mo>.</mo> <mo>.</mo> <mo>.</mo> <mo>,</mo> <mi>N</mi> <mo>-</mo> <mn>1</mn> <mo>)</mo> </mrow> <mo>-</mo> <mo>-</mo> <mo>-</mo> <mrow> <mo>(</mo> <mn>6</mn> <mo>)</mo> </mrow> </mrow> </math>
the time domain signal is changed into a time domain signal, the IDFT point number is more than N during oversampling, the high frequency part is set to zero, and the time domain signal is modulated and sent out after D/A (digital-to-analog conversion) is carried out on the time domain signal.
And thirdly, selecting the signals transmitted on the available sub-channels, then balancing the selected signals, converting the signals back to a time domain for judgment, and finally obtaining the transmitted data.
The receiving end receives the signal, and the time domain discrete signal after the CP is removed is:
<math> <mrow> <msup> <mi>r</mi> <mo>&prime;</mo> </msup> <mrow> <mo>(</mo> <mi>n</mi> <mo>)</mo> </mrow> <mo>=</mo> <msup> <mi>s</mi> <mo>&prime;</mo> </msup> <mrow> <mo>(</mo> <mi>n</mi> <mo>)</mo> </mrow> <mo>&CircleTimes;</mo> <mi>h</mi> <mrow> <mo>(</mo> <mi>n</mi> <mo>)</mo> </mrow> <mo>+</mo> <mi>w</mi> <mrow> <mo>(</mo> <mi>n</mi> <mo>)</mo> </mrow> <mo>,</mo> <mrow> <mo>(</mo> <mi>n</mi> <mo>=</mo> <mn>0,1</mn> <mo>,</mo> <mo>.</mo> <mo>.</mo> <mo>.</mo> <mo>,</mo> <mi>N</mi> <mo>-</mo> <mn>1</mn> <mo>)</mo> </mrow> <mo>-</mo> <mo>-</mo> <mo>-</mo> <mrow> <mo>(</mo> <mn>7</mn> <mo>)</mo> </mrow> </mrow> </math>
DFT of N points is carried out:
<math> <mrow> <msup> <mi>R</mi> <mo>&prime;</mo> </msup> <mrow> <mo>(</mo> <mi>k</mi> <mo>)</mo> </mrow> <mo>=</mo> <munderover> <mi>&Sigma;</mi> <mrow> <mi>n</mi> <mo>=</mo> <mn>0</mn> </mrow> <mrow> <mi>N</mi> <mo>-</mo> <mn>1</mn> </mrow> </munderover> <msup> <mi>r</mi> <mo>&prime;</mo> </msup> <mrow> <mo>(</mo> <mi>n</mi> <mo>)</mo> </mrow> <msup> <mi>e</mi> <mrow> <mo>-</mo> <mi>j</mi> <mfrac> <mrow> <mn>2</mn> <mi>&pi;</mi> </mrow> <mi>N</mi> </mfrac> <mi>nk</mi> </mrow> </msup> <mo>,</mo> <mrow> <mo>(</mo> <mi>k</mi> <mo>=</mo> <mn>0,1</mn> <mo>,</mo> <mo>.</mo> <mo>.</mo> <mo>.</mo> <mo>,</mo> <mi>N</mi> <mo>-</mo> <mn>1</mn> <mo>)</mo> </mrow> <mo>-</mo> <mo>-</mo> <mo>-</mo> <mrow> <mo>(</mo> <mn>8</mn> <mo>)</mo> </mrow> </mrow> </math>
and:
R′(k)=S′(k)H(k)+W(k),(k=0,1,…,N-1) (9)
this allows selection of the signal R (k) on the M available subchannels based on the subchannel label informationi) (i-0, 1, …, M-1), and then using the available subchannel parameters H (k) in the estimated channel state informationi) (i-0, 1, …, M-1), equalizing the selected signal; one of three equalization modes can be selected:
1. zero-forcing equalization is carried out on the data,
2. the minimum mean square error is balanced, and the minimum mean square error is balanced,
3. hybrid equalization, i.e. one part of the subchannels is equalized with zero-forcing and the other part of the subchannels is equalized with minimum mean square error;
zero forcing equalization is taken as an example for introduction:
S ~ ( k i ) = R ( k i ) H ( k i ) , ( i = 0,1 , . . . , M - 1 ) - - - ( 10 )
order to
S ~ ( i ) = S ~ ( k i ) , ( i = 0,1 , . . . , M - 1 ) - - - ( 11 )
IDFT to which M points are made:
<math> <mrow> <mover> <mi>s</mi> <mo>~</mo> </mover> <mrow> <mo>(</mo> <mi>n</mi> <mo>)</mo> </mrow> <mo>=</mo> <mfrac> <mn>1</mn> <mi>M</mi> </mfrac> <munderover> <mi>&Sigma;</mi> <mrow> <mi>i</mi> <mo>=</mo> <mn>0</mn> </mrow> <mrow> <mi>M</mi> <mo>-</mo> <mn>1</mn> </mrow> </munderover> <mover> <mi>S</mi> <mo>~</mo> </mover> <mrow> <mo>(</mo> <mi>i</mi> <mo>)</mo> </mrow> <msup> <mi>e</mi> <mrow> <mi>j</mi> <mfrac> <mrow> <mn>2</mn> <mi>&pi;</mi> </mrow> <mi>M</mi> </mfrac> <mi>ni</mi> </mrow> </msup> <mo>,</mo> <mrow> <mo>(</mo> <mi>n</mi> <mo>=</mo> <mn>0,1</mn> <mo>,</mo> <mo>.</mo> <mo>.</mo> <mo>.</mo> <mo>,</mo> <mi>M</mi> <mo>-</mo> <mn>1</mn> <mo>)</mo> </mrow> <mo>-</mo> <mo>-</mo> <mo>-</mo> <mrow> <mo>(</mo> <mn>12</mn> <mo>)</mo> </mrow> </mrow> </math>
the decision on this set of data can recover the original data.
The single carrier block transmission method of the frequency selection mode utilizes the channel state information and can avoid deep attenuation points for frequency selective fading channels, thereby obviously improving the error code performance of the system. Communication systems generally have certain performance requirements, and the system performance is determined by the signal-to-noise ratio after equalization, which is the ratio of the signal power and the noise power after equalization, and the signal-to-noise ratio loss caused by channel estimation errors and synchronization errors. And the signal-to-noise ratio after equalization, the signal-to-noise ratio loss caused by the channel estimation error and the synchronization error also determine the channel capacity of the system, so the modulation mode of the system is adaptively adjusted according to the signal-to-noise ratio after equalization, the signal-to-noise ratio loss caused by the channel estimation error and the synchronization error of the system and the performance requirements of the system, and the spectrum efficiency can be further improved.
Disclosure of the invention
Aiming at the single carrier block transmission method of the frequency selection mode, the invention provides a bit loading method which can make full use of the transmission power and improve the utilization rate of the system frequency spectrum based on the transmission method.
The bit loading method comprises the following steps:
(1) after the two parties of the transceiver establish communication, the receiving end gains | H (k) according to the number M of the selected available sub-channels and the amplitude of the sub-channelsi) I ═ 0, 1, …, M-1), yielding the equalized noise power, denoted as σ2Obtaining the equalized SNR of the system according to the received SNR and the equalization mode and recording the equalized SNR as the SNReq
(2) Determining different linear modulation modes adopted by the system according to the equalized signal-to-noise ratio of the system and the signal-to-noise ratio loss caused by channel estimation errors and synchronization errors, wherein data describing the different linear modulation modes are called modulation mode information, and the modulation mode information is transmitted to a transmitting end through a feedback channel;
(3) the sending end carries out symbol mapping according to the received modulation mode information to send signals;
(4) and the receiving end demodulates the signal according to the modulation mode information and judges.
The above steps are explained in detail below:
and (1) the receiving end calculates the equalized signal-to-noise ratio according to the sub-channel marking information, the received signal-to-noise ratio and the equalization mode.
The method for measuring and calculating the received signal-to-noise ratio can refer to relevant documents, and is not described herein. The equalization modes are different, and the equalized signal power and noise power are notSimilarly, the post-equalization signal-to-noise ratio of the system is also different, and when zero-forcing equalization is adopted, the method for calculating the post-equalization signal-to-noise ratio is as follows: setting the receiving end to select M available sub-channels according to the channel state information, the amplitude gain of these sub-channels is | H (k)i) (i ═ 0, 1, …, M-1), and white gaussian noise bilateral power spectral density in units of W/Hz
Figure C20051004205600071
The post-equalization noise power is, in units of W:
<math> <mrow> <msup> <mi>&sigma;</mi> <mn>2</mn> </msup> <mo>=</mo> <mi>E</mi> <mrow> <mo>(</mo> <munderover> <mi>&Sigma;</mi> <mrow> <mi>i</mi> <mo>=</mo> <mn>0</mn> </mrow> <mrow> <mi>M</mi> <mo>-</mo> <mn>1</mn> </mrow> </munderover> <msup> <mrow> <mo>|</mo> <mfrac> <mrow> <mi>N</mi> <mrow> <mo>(</mo> <msub> <mi>k</mi> <mi>i</mi> </msub> <mo>)</mo> </mrow> </mrow> <mrow> <mi>H</mi> <mrow> <mo>(</mo> <msub> <mi>k</mi> <mi>i</mi> </msub> <mo>)</mo> </mrow> </mrow> </mfrac> <mo>|</mo> </mrow> <mn>2</mn> </msup> <mo>)</mo> </mrow> <mo>=</mo> <mfrac> <msub> <mi>N</mi> <mn>0</mn> </msub> <mn>2</mn> </mfrac> <munderover> <mi>&Sigma;</mi> <mrow> <mi>i</mi> <mo>=</mo> <mn>0</mn> </mrow> <mrow> <mi>M</mi> <mo>-</mo> <mn>1</mn> </mrow> </munderover> <msup> <mrow> <mo>|</mo> <mfrac> <mn>1</mn> <mrow> <mi>H</mi> <mrow> <mo>(</mo> <msub> <mi>k</mi> <mi>i</mi> </msub> <mo>)</mo> </mrow> </mrow> </mfrac> <mo>|</mo> </mrow> <mn>2</mn> </msup> <mo>-</mo> <mo>-</mo> <mo>-</mo> <mrow> <mo>(</mo> <mn>13</mn> <mo>)</mo> </mrow> </mrow> </math>
N(ki) Is serial number kiNoise before equalization, k, on a subchanneliIs the serial number of the ith available sub-channel, and the equalized signal power is set as SeqThen the equalized signal-to-noise ratio of the system is:
<math> <mrow> <mi>SN</mi> <msub> <mi>R</mi> <mi>eq</mi> </msub> <mo>=</mo> <mfrac> <msub> <mi>S</mi> <mi>eq</mi> </msub> <msup> <mi>&sigma;</mi> <mn>2</mn> </msup> </mfrac> <mo>.</mo> <mo>-</mo> <mo>-</mo> <mo>-</mo> <mrow> <mo>(</mo> <mn>14</mn> <mo>)</mo> </mrow> </mrow> </math>
and (2) determining different linear modulation modes adopted by the system according to the equalized signal-to-noise ratio of the system, the signal-to-noise ratio loss caused by channel estimation errors and synchronization errors.
Through analysis, when channel estimation errors and synchronization errors are negligible, a frequency-selective single-carrier block transmission system in a multipath environment can be equivalent to a communication system passing through an ideal Gaussian channel after being equalized, and the signal-to-noise ratio after equalization is equivalent to the signal-to-noise ratio under the ideal Gaussian channel, so that the bit error rate of the system is determined by the signal-to-noise ratio after equalization, and the SNR is SNReqWhen considering the channel estimation error and the synchronization error, the loss of the signal-to-noise ratio can be estimated by measuring the bit error ratio performance under the actual received signal-to-noise ratio and calculating the signal-to-noise ratio required by the system without the channel estimation error and the synchronization error to reach the same bit error ratio, and the loss is recorded as the SNRlossThe method for calculating the SNR loss can refer to the relevant literature, and the error performance of the system is determined by the ratio of the SNR after equalization to the SNR loss, which is recorded as SNR ═ SNReq/SNRlossIn which the signal-to-noise ratio is lostlossThe calculation method of (2) can be referred to relevant documents;
if a modulation mode is adopted, the bit error rate BER required by the system is achievedreqThe required signal-to-noise ratio is generally higher than or lower than the SNR, and in order to fully utilize the power and improve the transmission rate, two or more modulation modes with different systems can be adopted. Because each time domain point of the system can only carry information of an integer bit, the number of the adopted modulation mode is generally 2kIn the frequency-selective single carrier block transmission system, a linear modulation scheme, such as QAM or MPSK, is commonly adopted, and two modulation schemes are described as follows:
under the condition of setting ideal Gauss channel, when adopting two system linear modulation modes, it can attain the same system error code requirement, i.e. error bit rate BERreqThe required signal-to-noise ratios are SNR respectivelyk(k is 1, 2, …), and the number of information bits carried by the two different binary linear modulation modes is k1And k2And k is2=k1+1. If it satisfies
SNRk≤SNR ≤SNRk+1,(k=1,2,…) (15)
Then: k is a radical of1=k (16)
The average mapping power refers to the average power corresponding to all constellation points when mapping is performed by adopting different modulation modes. Setting the average mapping power of the two selected linear modulation modes as S1And S2Then k is carried in each frame2The number of points of bit information is:
Figure C20051004205600081
Figure C20051004205600082
indicating a lower rounding. Then k is carried in each frame1The number of points of bit information is:
M1=M-M2 (18)
thus, each frame has M1One time domain point adoption
Figure C20051004205600083
Binary modulation, M2One time domain point adoption
Figure C20051004205600084
The binary modulation, in practical cases, leaves a certain margin in the power supplied by the system, so that the system can achieve the required relatively stable performance.
Will be adopted in M time domain points per frame
Figure C20051004205600085
Point number M of binary modulation mode1And the number of information bits k carried per point1And forming modulation mode information, and transmitting the modulation mode information and the sub-channel mark information to a transmitting end.
Step (3), the transmitting end performs symbol mapping according to the received modulation mode information, performs signal transformation according to the sub-channel mark information, and transmits a signal;
the transmitting end modulates the mode information k according to the feedback1,M1And the number M of available subchannels, where M can be obtained according to the subchannel flag information, and the modulation mapping can be performed according to the transmission signal power given by the system, and M is arbitrarily selected1A time domain point, each point is enabled to carry k1Bit information is carried out
Figure C20051004205600086
Binary modulation, M remaining2A time domain point, each point carrying k2Bit information is carried outBinary modulation, specific points being used
Figure C20051004205600088
Binary modulation, which ones adopt
Figure C20051004205600089
The binary modulation being defined by a convention or communication protocol used by the communicating parties, e.g. the first M of a frame of data may be selected1One time domain point adoptionBinary modulation, post M2One time domain point sampling
Figure C200510042056000811
Carrying out binary modulation; distributing the total transmission power of each frame of the transmitter to M time domain points according to the average mapping power ratio of different modulation modes, and then transmitting signals, for example, the former M1One time domain point adoption
Figure C20051004205600091
Binary modulation, post M2One time domain point adoption
Figure C20051004205600092
Binary modulation, frontM1Averaging the power allocated to each point in time domain by M2The average power ratio of each time domain point is
Figure C20051004205600093
And the power allocated to the time domain points adopting the same modulation mode in each frame is the same. After the symbol mapping is completed, signal transformation is carried out according to the sub-channel mark information by the frequency-selecting single-carrier block transmission method mentioned in the background technology, and signals are sent.
And (4) demodulating the signal by the receiving end according to the modulation mode information.
The receiving end selects the signal on the available sub-channel according to the sub-channel mark information, the signal is changed back to the time domain after equalization, the signal is demodulated and judged according to the modulation mode adopted by different time domain points to obtain correct transmission data, the demodulation method is similar to the signal mapping carried out by the sending end, which is the inverse process of the signal mapping, if according to the example of the step (3), the former M is the signal when demodulating1One time domain point for
Figure C20051004205600094
Carry the demodulation decision in the system, then M2One time domain point for
Figure C20051004205600095
And carrying out binary demodulation judgment.
The single carrier block transmission method based on the frequency selection mode adaptively loads the information bits in the time domain, and can more fully utilize the signal transmitting power. Under the condition of ensuring the stable error rate, the transmission code rate can be adjusted in a self-adaptive manner according to the received signal-to-noise ratio and the balance mode as well as the signal-to-noise ratio loss caused by the channel estimation error and the synchronization error. The self-adaptive method provided by the invention is slightly poorer than the traditional self-adaptive OFDM system in overall performance, but is far lower than the traditional self-adaptive OFDM system in the aspect of implementation complexity.
Under the condition of ensuring certain system performance, the invention has better spectrum efficiency and power efficiency than the prior SC-FDE and OFDM systems, and the added complexity of the whole system is small.
(IV) description of the drawings
The figure is a block diagram of a system implementing the proposed method of the invention.
In the figure: 1. the system comprises a source module, 2, a symbol mapping module, 3, an FFT module (M point), 4, a signal spectrum transformation module, 5, an IFFT module (N point), 6, a Cyclic Prefix (CP) adding module, 7, a D/A module, 8, an intermediate frequency and radio frequency modulation module, 9, a channel, 10, a radio frequency and intermediate frequency demodulation module, 11, an A/D module, 12, a CP removing module, 13, an FFT module (N point), 14, a signal spectrum inverse transformation module, 15, an equalization module, 16, an IFFT module (M point), 17, a judgment module, 18, a synchronization module, 19, a channel estimation module, 20, a modulation mode determination module, 21, a reverse channel estimation module
(V) detailed description of the preferred embodiments
Example (b):
the attached drawings show a system block diagram for realizing the method provided by the invention, and the functions of all modules are as follows:
the information source module 1: data to be transmitted is generated.
The symbol mapping module 2: and selecting modulation modes (QAM or MPSK) with different systems according to the modulation mode information returned by the reverse channel 21, and mapping the data generated by the information source to the corresponding points of the constellation diagram.
M-point FFT module 3: and transforming the M mapped signals of each frame to a frequency domain to obtain M point frequency domain signals of the signals.
The signal spectrum transformation module 4: according to the sub-channel marking information sent back by the receiving end through the reverse channel 21, the M-point frequency domain signal output by the module 3 is placed on the corresponding spectrum points of the M available sub-channels, and the corresponding spectrum points of the forbidden sub-channels are set to zero or non-information data is filled, so that a frame of new SC-FDE frequency domain signal with N points is obtained. This module needs to be programmed according to the method described in the invention patent mentioned in the background of the invention (patent application No. 200410036439.6), and is implemented by a general-purpose digital signal processing chip.
N-point IFFT module 5: and transforming the newly obtained frequency domain signal to the time domain.
And a CP adding module 6: and adding a cyclic prefix to each frame of obtained data.
D/A module 7: the digital signal is converted into an analog signal.
Intermediate frequency and radio frequency modulation module 8: if the system is used in a wireless environment, radio frequency modulation of the signal is required to transmit to the antenna. Sometimes, the signal needs to be modulated to an intermediate frequency for intermediate frequency amplification, then radio frequency modulation, and finally the modulated signal is sent to an antenna for transmission. If the system is used in a wired environment (e.g., xDSL), no rf modulation is required, no antenna is required to transmit the signal, and the signal spectrum is shifted outside the voice channel band to ensure that data is transmitted without affecting voice transmission.
Channel 9: a wired channel or a wireless channel that transmits signals.
The synchronization module 18: various synchronous data needed by the system are obtained through a parameter estimation method (such as blind estimation and auxiliary data-based estimation). The synchronization module sends the frequency synchronization data to the radio frequency and intermediate frequency demodulation module 10; sending the sampling rate synchronization data to the analog-to-digital conversion module 11; the timing synchronization data is sent to the de-CP module 12.
Radio frequency and intermediate frequency demodulation module 10: in a wireless environment, the frequency spectrum of a signal received by a receiving antenna is shifted from a radio frequency or an intermediate frequency to a low frequency. Frequency offset caused during signal transmission needs to be corrected by frequency synchronization data before demodulation.
The A/D module 11: the demodulated analog signal is converted into a digital signal. The analog signal needs to be sampled by the a/D, and the crystal oscillator providing the clock signal needs to have the same frequency as the crystal oscillator of the D/a module of the transmitter, otherwise, a sampling rate error may result. The sampling rate synchronization is performed before the a/D.
The CP removing module 12: the cyclic prefix is removed. There is a problem in determining when a frame of data starts, and therefore timing synchronization is required before the CP is removed.
N-point FFT module 13: the CP-removed signal is transformed into the frequency domain.
The channel estimation module 19: similar to synchronization, CSI is also obtained by parameter estimation, typically blind channel estimation and auxiliary data based channel estimation. After estimating the CSI, selecting available sub-channels, and sending the available sub-channel parameters to the equalization module 15; meanwhile, according to whether the channel is available, marking with 1 bit information (0 or 1) to form sub-channel marking information, sending the sub-channel marking information to the signal spectrum inverse transformation module 14 and the reverse channel 21 at the same time, and sending the sub-channel marking information back to the signal spectrum transformation module 4 of the sending end through the reverse channel; and passes the available subchannel status information to the modulation scheme determination module 20. This module needs to be programmed according to the method described in the invention patent mentioned in the background of the invention (patent application No. 200410036439.6), and is implemented by a general-purpose digital signal processing chip.
The signal spectrum inverse transformation module 14: according to the subchannel flag information sent by the channel estimation module 19, M-point frequency domain signals carried by available subchannels in the received signal are found out. This module needs to be programmed according to the method described in the invention patent mentioned in the background of the invention (patent application No. 200410036439.6), and is implemented by a general-purpose digital signal processing chip.
The equalization module 15: the signal selected by the inverse signal spectrum transform module 14 is equalized by the available sub-channel parameters from the channel estimation module 19. The equalization mode can select one of the following three equalization modes: zero-forcing equalization, minimum mean square error equalization, hybrid equalization (i.e., one portion of subchannels equalized with zero-forcing and another portion of subchannels equalized with minimum mean square error).
M-point IFFT module 16: and transforming the M frequency domain signals of the equalized signals to a time domain.
A decision module 17: and finishing the judgment of the time domain signal according to the modulation mode information transmitted by the constellation diagram and modulation mode determining module 20.
Modulation scheme determination module 20: the signal-to-noise ratio after equalization of the system is calculated according to the state information of the available sub-channels transmitted by the channel estimation module 19 and the power of the transmitted signal provided by the transmitting end, the modulation mode adopted by each frame is determined according to the signal-to-noise ratio after equalization, the signal-to-noise ratio of the channel estimation error and the signal-to-noise ratio of the synchronous error loss, the modulation mode information is transmitted to the decision module 17, and is transmitted to the mapping module 2 of the transmitting end through the reverse channel 21. The module needs to be programmed according to the method introduced by the invention and is realized by a general digital signal processing chip.
Reverse channel 21: and transmitting the sub-channel mark information and the modulation mode information back to the transmitting end.
Simulation parameters of this embodiment:
simulation environment: matlab7.0
Total number of subchannels: 256 of N
Number of available subchannels, i.e. number of SC-FDE data symbols per frame: and M is 208.
Modulation mode: QAM
CP Length: 32
The controlled error rate is as follows: BERreq=10-4
Simulating the selected signal-to-noise ratio range: SNR is 9: 30(dB)
Loss of signal-to-noise ratio: SNRloss0(dB) (i.e. neglecting the loss of signal-to-noise ratio due to channel estimation error and synchronization error)
The channel model employed in this embodiment is a sample of the SUI-5 channel (one of the test channels proposed in the IEEE802.16 standard).
The first column of the attached table below gives the received signal-to-noise ratio (in dB) provided by the system; the second column and the third column are the bit number and the corresponding bit error rate of each frame signal transmission corresponding to different received signal-to-noise ratios when a bit loading method is adopted; the fourth column and the fifth column are the bit number and the corresponding bit error rate of each frame signal transmission corresponding to different receiving signal-to-noise ratios when a bit loading method is not adopted. It can be seen from the table that under the condition of the same received signal-to-noise ratio, the bit error rate of the system without adopting the bit loading method is 10 higher than the required bit error rate-4The bit loading method is much lower, but the transmission rate is far less than that of a system adopting a bit loading method with relatively stable bit error rate. This shows that the bit loading method can greatly improve the spectral efficiency under the condition of controlling the bit error rate to be relatively stable.
Attached table:
wherein "- -" indicates a bit error rate of less than 10-7
To avoid confusion, some of the terms mentioned in this specification are to be interpreted as follows:
1. signal-to-noise ratio after equalization: the ratio of signal power to noise power after equalization.
2. Average mapping power: and when mapping is carried out by different modulation modes, the average power of all different constellation points is corresponded.
3. Symbol: refers to data in which information bits are modulation mapped (also referred to as symbol mapped). Typically a complex number where the real and imaginary parts are integers.
4. A frame signal: for OFDM, a frame signal refers to N symbols that are IFFT-transformed at the transmitting end and N symbols that are FFT-transformed after CP is removed at the receiving end. For SC-FDE, a frame of signal refers to N information symbols between two adjacent CPs at the transmitting end, and refers to N symbols that are FFT-transformed after the CPs are removed at the receiving end. For the SC-FDE system realized by the method provided by the invention, a frame of signal refers to M symbols for FFT transformation at a transmitting end and refers to M symbols for IFFT transformation after equalization at a receiving end.
5. Sub-channel: for OFDM, SC-FDE baseband signals, a subchannel refers to a frequency point after FFT at the receiving end. For a radio frequency channel, a subchannel refers to a segment of the frequency spectrum of the radio frequency channel.
6. Signal-to-noise ratio: the ratio of signal power to noise power, wherein the signal-to-noise ratio mentioned in the summary of the invention is not logarithmic and has no unit; the signal-to-noise ratio mentioned in the examples is a logarithmic signal-to-noise ratio in dB.

Claims (3)

1.A bit loading method in a frequency-selective single carrier block transmission system is characterized in that:
the method comprises the following steps:
(1) after the two parties of the transceiver establish communication, the receiving end gains | H (k) according to the number M of the selected available sub-channels and the amplitude of the sub-channelsi) I 0, 1, …, M-1, and the equalized noise power is obtained, denoted as σ2Obtaining the equalized SNR of the system according to the received SNR and the equalization mode and recording the equalized SNR as the SNReq
(2) Determining different linear modulation modes adopted by the system according to the equalized signal-to-noise ratio of the system and the signal-to-noise ratio loss caused by channel estimation errors and synchronization errors, wherein data describing the different linear modulation modes are called modulation mode information, and the modulation mode information is transmitted to a transmitting end through a feedback channel;
(3) the sending end carries out symbol mapping according to the received modulation mode information to send signals;
(4) and the receiving end demodulates the signal according to the modulation mode information and judges.
2. The bit loading method in a frequency selective single carrier block transmission system according to claim 1, characterized in that: the step (1) is realized by adopting the following method:
when zero-forcing equalization is adopted, the method for calculating the signal-to-noise ratio after equalization is as follows: setting the receiving end to select M available sub-channels according to the channel state information, the amplitude gain of these sub-channels is | H (k)i) I is 0, 1, …, M-1, and white gaussian noise bilateral power spectral density in W/HzThe post-equalization noise power is in units of W:
<math> <mrow> <msup> <mi>&sigma;</mi> <mn>2</mn> </msup> <mo>=</mo> <mi>E</mi> <mrow> <mo>(</mo> <munderover> <mi>&Sigma;</mi> <mrow> <mi>i</mi> <mo>=</mo> <mn>0</mn> </mrow> <mrow> <mi>M</mi> <mo>-</mo> <mn>1</mn> </mrow> </munderover> <msup> <mrow> <mo>|</mo> <mfrac> <mrow> <mi>N</mi> <mrow> <mo>(</mo> <msub> <mi>k</mi> <mi>i</mi> </msub> <mo>)</mo> </mrow> </mrow> <mrow> <mi>H</mi> <mrow> <mo>(</mo> <msub> <mi>k</mi> <mi>i</mi> </msub> <mo>)</mo> </mrow> </mrow> </mfrac> <mo>|</mo> </mrow> <mn>2</mn> </msup> <mo>)</mo> </mrow> <mo>=</mo> <mfrac> <msub> <mi>N</mi> <mn>0</mn> </msub> <mn>2</mn> </mfrac> <munderover> <mi>&Sigma;</mi> <mrow> <mi>i</mi> <mo>=</mo> <mn>0</mn> </mrow> <mrow> <mi>M</mi> <mo>-</mo> <mn>1</mn> </mrow> </munderover> <msup> <mrow> <mo>|</mo> <mfrac> <mn>1</mn> <mrow> <mi>H</mi> <mrow> <mo>(</mo> <msub> <mi>k</mi> <mi>i</mi> </msub> <mo>)</mo> </mrow> </mrow> </mfrac> <mo>|</mo> </mrow> <mn>2</mn> </msup> </mrow> </math>
N(ki) Is serial number kiNoise before equalization, k, on a subchanneliIs the serial number of the ith available sub-channel, and the equalized signal power is set as SeqThen the equalized signal-to-noise ratio of the system is:
<math> <mrow> <msub> <mi>SNR</mi> <mi>eq</mi> </msub> <mo>=</mo> <mfrac> <msub> <mi>S</mi> <mi>eq</mi> </msub> <msup> <mi>&sigma;</mi> <mn>2</mn> </msup> </mfrac> <mo>.</mo> </mrow> </math>
3. the bit loading method in a frequency selective single carrier block transmission system according to claim 1, characterized in that: the step (2) is realized by adopting the following method:
when the channel estimation error and the synchronization error can be ignored, the frequency-selecting single-carrier block transmission system in the multipath environment is equivalent to a communication system passing through an ideal Gaussian channel after being equalized, and the signal-to-noise ratio after equalization is equivalent to the signal-to-noise ratio under the ideal Gaussian channel, so the error rate of the system is determined by the signal-to-noise ratio after equalization, and the signal-to-noise ratio is equal to the SNReq(ii) a When considering the channel estimation error and the synchronization error, the signal-to-noise ratio loss can be estimated by measuring the bit error ratio performance under the actual received signal-to-noise ratio and calculating the signal-to-noise ratio required by the system without the channel estimation error and the synchronization error to reach the same bit error ratio, and is recorded as the SNRlossAt this time, the ratio of the SNR after equalization to the SNR loss is used to determine the system error performance, and is recorded as SNR ═ SNReq/SNRloss
In order to fully utilize power and improve transmission rate, two or more than two modulation modes with different systems are adopted, and each subchannel of the system can only carry information of integer bits, so that the system number of the adopted modulation mode is 2kAnd k is 1, 2, …, in the frequency-selective single-carrier block transmission system, a linear modulation mode is adopted, and when two modulation modes are adopted:
under the condition of setting ideal Gauss channel, when adopting two system linear modulation modes, it can attain the same system error code requirement, i.e. error bit rate BERreqThe required signal-to-noise ratios are SNR respectivelykK is 1, 2, …, and the number of information bits carried by the two selected linear modulation schemes with different systems is k1And k2And k is2=k1+ 1; if it satisfies
SNRk≤SNR≤SNRk+1,k=1,2,…
Then: k is a radical of1=k
The average mapping power refers to the average power corresponding to all constellation points when mapping is performed by adopting different modulation modes, and the average mapping powers of the two selected linear modulation modes are respectively set as S1And S2Then k is carried in each frame2The number of subchannels of the bit information is:
Figure C2005100420560003C1
represents lower rounding; then k is carried in each frame1The number of subchannels of the bit information is:
M1=M-M2
thus, each frame has M1Sub-channel adoption
Figure C2005100420560003C3
Binary modulation, M2Sub-channel adoption
Figure C2005100420560003C4
In practical situations, a certain margin is left for the power provided by the system, so that the system can achieve the required relatively stable performance; will be employed in M sub-channels per frame
Figure C2005100420560003C5
Number of subchannels M of binary modulation scheme1And the number of information bits k carried by each subchannel1And forming modulation mode information, and transmitting the modulation mode information and the sub-channel mark information to a transmitting end.
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