Presentation on theme: "TRANSMISSION OF INFORMATION"— Presentation transcript:
1TRANSMISSION OF INFORMATION David Falconer and Halim YanikomerogluDept. of Systems and Computer EngineeringCarleton University
2Topics to be CoveredAnalog (continuous time, continuous amplitude) signalsPower spectral density and bandwidthAnalog to digital: PCM (pulse code modulation)Digital transmission
3Digital and Analog Signals Some signals (like speech and video) are inherently analog; some (like computer data) are inherently digital.However, both analog and digital signals can be represented and transmitted digitally.Advantages of digital:Reduced sensitivity to line noise, temp. drift, etc.Low cost digital VLSI for switching and transmission.Lower maintenance costs than analog.Uniformity in carrying voice, SMS, , data, video, etc. (a bit is a bit)Better encryption.
4Power Spectral Density Power spectrum (power spectral density) describes how the average power is distributed with respect to frequency.Deterministic signals Fourier transformRandom signals Power spectral densityA statistical representation for all random signals in a particular application
5Power Spectrum of Analog Signals Analog (continuous-time, continuous-amplitude) signals (like speech) have a certain bandwidth. Their power spectrum (power spectral density) describes how their average power is distributed with respect to frequency.Powerspectraldensity(watts/Hz)“High-fidelity speechTelephone speech(limited by filtering)Bandwidth
6Power Spectrum of Analog Signals Source: Wikipedia
7BandwidthFor random signals, bandwidth is determined from the power spectral density.Bandwidth is determined only from the +ve frequencies.There are different bandwidth definitionsAbsolute bandwidthY% bandwidth (for instance, 99%)X-dB bandwidth (for instance, 3-dB)Null-to-null bandwidth…
13Sampling an Analog Signal Sampling theorem: The original analog signal can be reconstructed if it is sampled at a rate at least twice its bandwidth.Reconstruction is by filtering samples with a low pass filter.Sampling Samples Reconstruction
14Pulse Code Modulation (PCM) PCM is a method used to digitally represent sampled analog signals. It is the standard form of digital audio in computers, Compact Discs, digital telephony and other digital audio applications.PCM signal is developed by three steps: sampling, quantizing and encoding.Quantizing noise is reduced by using variable sized steps. It is independent of line length.s(t)s(n)FilterSample at t=n Quantize Encode
15Pulse Code Modulation (PCM) Sampling and quantization of a signal (red) for 4-bit PCMThe PCM process is commonly implemented on a single integrated circuit and is generally referred to as an analog-to-digital converter (ADC)
16Standard PCM in Wired Telephony Voice circuit bandwidth is 3400 Hz.Sampling rate is 8 KHz (samples are 125 s apart).Each sample is quantized to one of 256 levels.Each quantized sample is coded into a 8-bit word.The 8-bit words are transmitted serially (one bit at a time) over a digital transmission channel. The bit rate is 8x8,000 = 64 Kb/s.The bits are regenerated at digital repeaters.The received words are decoded back to quantized samples, and filtered to reconstruct the analog signal.
17Quantization LPMC: Uncompressed Uniform (Linear PCM: LPCM) NonuniformOutput signalOutput signalInput signalInput signalThe more steps (levels) the less quantization noise. Nonuniform quantization(e.g. -law) allows a larger dynamic range (important for speech).LPMC: UncompressedNonuniform quantization: Introduces compression
18-Law Quantization and Coding Standardized in North America.Based on a logarithmic non-uniform quantizer.Range of amplitudes divided into 8 segments, each segment with 16 uniformly spaced levels. Segment i is double the width of segment i-1.8 bit word: 1 bit for sign, 3 bits identify segment, 4 bits identify level within segment.Can show for n-bit word, signal to quantization noise ratio is approximately 6n-10 [dB]; e.g., 38 dB for n=8 bits.Most of the rest of the world uses a related logarithmic non-uniformity, called A-law.
19Variants of PCM (Form of Compression) Differential PCM (DPCM) encodes the PCM values as differences between the current and the predicted value. An algorithm predicts the next sample based on the previous samples, and the encoder stores only the difference between this prediction and the actual value. If the prediction is reasonable, fewer bits can be used to represent the same information. For audio, this type of encoding reduces the number of bits required per sample by about 25% compared to PCM. Adaptive DPCM (ADPCM) is a variant of DPCM that varies the size of the quantization step, to allow further reduction of the required bandwidth for a given signal-to-noise ratio. Delta Modulation is a form of DPCM which uses one bit per sample.
20Adaptive Differential PCM (ADPCM) Allows coding with a lower bit rate (with same fidelity) for speech, based on predicting the next sample; e.g., 8 or 16 or 32 Kb/s.More circuits accommodated in the same transmission bandwidth.Coder: Decoder:+Quant.+PredictorPredictor
21PCM StandardsG.711 is an ITU-T standard for audio companding. It is primarily used in telephony. The standard was released for usage in Its formal name is Pulse Code Modulation (PCM) of voice frequencies. G.711 uses a sampling rate of 8,000 samples per second. Non-uniform (logarithmic) quantization with 8 bits is used to represent each sample, resulting in a 64 kbit/s bit rate. G is an extension to G.711, published as ITU-T Recommendation G in March Its formal name is Wideband embedded extension for G.711 pulse code modulation. G.711.1, allows the addition of narrowband and/or wideband (16000 samples/s) enhancements, each at 25 % of the bitrate of the (included) base G.711 bitstream, leading to data rates of 64, 80 or 96 kbit/s. G is compatible with G.711 at 64 kbit/s, hence an efficient deployment in existing G.711-based voice over IP (VoIP) infrastructures is foreseen.
22PCM StandardsG.726 is an ITU-T ADPCM speech codec standard covering the transmission of voice at rates of 16, 24, 32, and 40 kbit/s (1990). The most commonly used mode is 32 kbit/s, which doubles the usable network capacity by using half the rate of G.711. It is primarily used on international trunks in the phone network. The principal application of 24 and 16 kbit/s channels is for overload channels carrying voice in digital circuit multiplication equipment (DCME). It also is the standard codec used in DECT wireless phone systems and is used on some Canon cameras. Sampling frequency 8 kHz. 16 kbit/s, 24 kbit/s, 32 kbit/s, 40 kbit/s bit rates available. Testing under ideal conditions yields Mean Opinion Scores of 4.30 for G.726 (32 kbit/s), compared to 4.45 for G.711 (µ-law)
23PCM StandardsAudio Interchange File Format (AIFF) is an audio file format standard used for storing sound data for personal computers and other electronic audio devices. The audio data in a standard AIFF file is uncompressed pulse-code modulation (PCM). There is also a compressed variant of AIFF known as AIFF-C or AIFC, with various defined compression codecs. Like any non-compressed, lossless format, it uses much more disk space than MP3—about 10MB for one minute of stereo audio at a sample rate of 44.1 kHz and a sample size of 16 bits. Developed by Apple Inc. Initial release 21 January 1988.
24PCM StandardsMPEG-1 or MPEG-2 Audio Layer III, more commonly referred to as MP3, is a patented digital audio encoding format using a form of lossy data compression. It is a common audio format for consumer audio storage, as well as a de facto standard of digital audio compression for the transfer and playback of music on digital audio players. Initial release: MP3 is an audio-specific format that was designed by the Moving Picture Experts Group (MPEG) as part of its MPEG-1 standard and later extended in MPEG-2 standard. The use in MP3 of a lossy compression algorithm is designed to greatly reduce the amount of data required to represent the audio recording and still sound like a faithful reproduction of the original uncompressed audio for most listeners. An MP3 file that is created using the setting of 128 kbit/s will result in a file that is about 1/11 the size of the CD file created from the original audio source. An MP3 file can also be constructed at higher or lower bit rates, with higher or lower resulting quality. The compression works by reducing accuracy of certain parts of sound that are considered to be beyond the auditory resolution ability of most people. This method is commonly referred to as perceptual coding. It uses psychoacoustic models to discard or reduce precision of components less audible to human hearing, and then records the remaining information in an efficient manner.
25PCM StandardsSeveral bit rates are specified in the MPEG-1 Audio Layer III standard: 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256 and 320 kbit/s, and the available sampling frequencies are 32, 44.1 and 48 kHz. Additional extensions were defined in MPEG-2 Audio Layer III: bit rates 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160 kbit/s and sampling frequencies 16, and 24 kHz. A sample rate of 44.1 kHz is almost always used, because this is also used for CD audio, the main source used for creating MP3 files. A greater variety of bit rates are used on the Internet. The rate of 128 kbit/s is commonly used, at a compression ratio of 11:1, offering adequate audio quality in a relatively small space. As Internet bandwidth availability and hard drive sizes have increased, higher bit rates up to 320 kbit/s are widespread. Uncompressed audio as stored on an audio-CD has a bit rate of 1,411.2 kbit/s, so the bitrates 128, 160 and 192 kbit/s represent compression ratios of approximately 11:1, 9:1 and 7:1 respectively.
26DS1 Format (-Law Countries) 24S bit24 PCM code words, each representing 1 sample8 bits per code word193 bits in 125 s(1.544 Mb/s)DS1DS0Hierarchical Multiplexing:DS1: 24 DS0DS3: 28 DS1
27Regenerative Repeater Amplifier/equalizerRegeneratorStructure of a regenerativerepeater:Timing circuitBy appropriate repeater design and inter-repeater spacing, the effect ofoccasional bit errors due to noise can be controlled. Received signal quality isessentially independent of distance.
29Multilevel Transmission Binary:L=24-level:L=4T T T TBit rate =Bandwidth proportional to 1/T for NRZ signals
30Bandwidth Required for Digital Transmission required bandwidth is approximately(bit rate)/(log2L) for L-level transmission.more levels less bandwidth, but greater sensitivity to noise.Examples:64 Kb/s PCM requires about 64 KHz for binary transmission, 32 KHz for 4-level transmission.14.4 Kb/s modem uses a symbol rate 1/T=2400 Hz, and the equivalent of L=32.
31Channel Capacity Shannon channel capacity formula: Highest possible transmission bit rate R, for reliable communication in a given bandwidth W Hz, with given signal to noise ratio, SNR, isR=Wlog2(1+SNR) bits/sR/W = SNR [dB] bits/s/Hz (for high SNR)Assumptions and qualifications:Gaussian distributed noise added to the signal by the channel, highly complex modulation, coding and decoding methods.In typical practical situations, the above formula may be roughly modified by dividing SNR by a factor of about 5 to 10.Note: For any x, log2(x)=(1/log10(2)) log10(x)=3.32 log10(x)e.g. for SNR=10 dB, R/W=3.45 bit/s per HzFor high SNR, R/W 0.332 X (SNR expressed in dB)e. g. for SNR=30 dB, R/W 10 bits/s per Hz.
32Fundamental Limits in Digital Data Rates Mobile device for everything ?Mobile device for everything3GGbps2GMbps1GKbps2010 – will be the decade of LTE2020 -> - will be 5th G what will it hold?Not entirely sure but – what is sure is that Between Now and then-Cell size will definitely shrink – and consequentlyCell count will increaseAll in an order of magnitudeDevices will undoubtedly expand exponentiallyData M2M will explode – And this will inevitably lead to a concatenation of growth trendsThe growing need for information and interaction will spur a number of Mega-trends ……bpsAMPS19801990200020102020Time32
33Information Theory and Digital Communications Ralph V.L. Hartley1888 – 1970Harry Nyquist1889 – 1976Norbert Wiener1894 – 1964Claude Shannon1916 – 2001Emre Telatar1964 –Gerard J. Foschini1940 –
34Fundamental Limits in Digital Data Rates RBS: Data rate (speed) of a wireless base station (access point)W: BandwidthSNR: Signal-to-noise ratio at the receiverSE: Spectral efficiency = log2(1+SNR)n: Min (# of transmit antennas, # of receive antennas)None of the three variables (W, SE, n) scales well!Ex 1: n = 2, W = 10 MHz, log(1+SNR) = RBS = 80 MbpsEx 2: n = 8, W = 100 MHz, log(1+SNR) = 4.5 RBS = 3.6 Gbps(Cellular 4th generation LTE-Advanced)RBS = n x W x SE = n x W x log2(1+SNR)34
35Fundamental Limits in Digital Data Rates Rnetwork: Network rateK: # of BSs in the networkFundamental dynamics:4 basic factors that impact network rate: K, n, W, SEIncreasing base station rate: Not easy! (neither of n, W, SE scales well)Increasing network rate: Possible! (by adding more base stations)Rnetwork = K x n x W x log2(1+SNR)35
36SummaryAll information signals can be represented, switched, stored and transmitted digitally.We have discussed PCM systems and their key elements:samplingquantizingcodingdigital transmissionWe have discussed the related concepts of:the telephone setbandwidththe sampling theoremsignal to quantization noise ratiochannel capacity.