Telecommunications Network Engineering: PCM Coding, Speech Methods, and Power Levels

Structure of the Eight-Bit Binary Word in European PCM (A-law)

  • The structure of the 8-bit binary word used in European Pulse Code Modulation (PCM) coding, known as the A-law, is defined by specific bit functions:

    • Bit 1 (Most Significant Bit - MSB): This is the first bit in the sequence and indicates the polarity of the sample.

      • A value of 11 represents positive polarity.

      • A value of 00 represents negative polarity.

      • The sample value of zero can produce two distinct code words depending on whether it is interpreted as having positive or negative polarity.

    • Bits 2, 3, and 4: These three bits define the specific segment where the sample value is located.

      • Segments indexed as 000000 and 001001 together form a linear curve designated for low-level positive or negative samples.

      • The complete A-law curve consists of 1313 linear sections (as illustrated in Figure 3.14).

    • Bits 5, 6, 7, and 8 (Least Significant Bits - LSBs): these bits reveal the specific quantized value of the sample within its assigned segment.

      • Each segment is divided linearly into 1616 distinct values or quantum levels.

  • Coding Relationship and Compression:

    • The relationship between signal values and binary words is nonlinear, combining nonlinear compression with linear coding.

    • The vertical axis is linear; each binary word corresponds to a quantum level at an equal distance from the others.

    • Linear quantizing is performed on the compressed signal.

    • The relationship between the vertical and horizontal axes follows a logarithmic scale according to equation (3.9).

    • To reduce quantizing noise at low signal levels, half of the quantum levels are dedicated to signal levels smaller than 6.25%6.25\% (or 116\frac{1}{16}) of the maximum value.

  • Bit Inversion and Synchronization:

    • Following the encoding process, every other bit of the resulting code words is inverted prior to multiplexing.

    • The purpose of this inversion is to "mix" the digital signal, facilitating easier timing for line systems and equipment interfaces.

    • For instance, a small negative value producing the code word 0000000000000000 would be inverted to 0101010101010101.

    • Without this inversion, transmitting continuous zeros could lead to synchronization difficulties for the receiver.

PCM Encoder and Decoder Standards

  • Historical Context: PCM coding schemes for digital voice communications were standardized by the CCITT (now ITU-T) in the early 1970s based on contemporary technology.

  • Regional Variations:

    • The European standard (A-law) differs slightly from the American standard (u-law), necessitating conversion equipment for trans-Atlantic or European-to-Japanese communications.

    • The majority of countries worldwide utilize the European A-law standard.

The PCM Encoder Process

  • Filtering: Before encoding, the analog signal passes through a lowpass filter (LPF) limiting the frequency band to 300300 to 3,400Hz3,400\,Hz. This bandwidth is sufficient for human voice recognition and speaker identification.

  • Sampling Theorem Compliance: This filtering ensures the sampling theorem is satisfied by removing frequencies higher than half the sampling frequency (fs/2f_s/2).

  • Encoding Steps:

    • The signal is sampled at an 8kHz8\,kHz sampling frequency.

    • A sample-and-hold (S/H) circuit captures the values.

    • A 256256-level quantizer and encoder convert samples into 8-bit nonlinearly coded words.

    • Parallel words are converted to a serial form via a parallel-to-serial converter.

  • Multiplexing and Data Rates:

    • Individual 64Kbps64\,Kbps serial channels are multiplexed using Time Division Multiplexing (TDM).

    • According to the European standard, 3030 voice channels are combined into a 2,048Kbps2,048\,Kbps (or 2Mbps2\,Mbps) primary rate signal.

    • Digital exchanges often build 2Mbps2\,Mbps streams with 3030 PCM-coded subscriber interfaces for internal transmission.

  • United States DS1 System:

    • The data rate is 1.544Mbps1.544\,Mbps instead of 2.048Mbps2.048\,Mbps.

    • Each frame contains 2424 speech channels plus a single framing bit.

    • The math for the data rate is defined as:     8,000×(8×24+1)=1,544,000bps=1.544Mbps8,000 \times (8 \times 24 + 1) = 1,544,000\,bps = 1.544\,Mbps

The PCM Decoder Process

  • Demultiplexing: The receiver uses a TDM demultiplexer to separate the 2,048Kbps2,048\,Kbps stream back into individual 64Kbps64\,Kbps channels.

  • Reconstruction:

    • The serial signal is converted back to 88-bit parallel sample values.

    • A decoder reconstructs the sample pulses.

    • A lowpass filter (LPF) smooths the pulses to recreate a voice signal resembling the original analog input.

Advanced Speech-Coding Methods

  • Efficiency Definition: "More efficient" indicates achieving better quality at the same data rate or maintaining equal quality at a lower data rate.

  • Categories of Coding:

    • Waveform Coding (e.g., PCM): Transmits information describing the signal waveform in the time domain. It adds minimal delay and provides high quality but requires higher data rates.

    • Voice Coding (Vocoders): Utilizes human voice characteristics by modeling segments (e.g., 50ms50\,ms) and sending model identification codes. Vocoders introduce analytical delay and may split voice into frequency components.

    • Hybrid Coders: Combines waveform and voice coding principles for a compromise between quality and data rate.

  • Adaptive PCM (APCM):

    • Transmits signal strength information periodically alongside sample values.

    • Fewer bits are required for samples, as they define the quantum level within a variable scale.

    • Keeping the SQR (Signal-to-Quantizing Ratio) high over a wide range of signal levels.

  • Differential PCM (DPCM):

    • Only the difference between a current sample and the previous value is encoded.

    • Since differences are typically small, fewer bits are needed (e.g., 44 bits instead of 55 bits).

    • Prediction: Can use multiple preceding samples (typically up to three) to predict the next value; the encoder only transmits the difference between the prediction and the actual sample.

  • Delta Modulation (DM):

    • A simplified DPCM that transmits a binary 11 if the sample increases and a binary 00 if it decreases.

    • Continuous Variable Slope Delta (CVSD) Modulation: Uses large quantizing steps for steep slopes and small steps for minimal changes. Used in Bluetooth and military applications.

  • Adaptive DPCM (ADPCM):

    • Combines APCM and DPCM, adapting both the predictor and the quantizer.

    • Quantization Scaling: If the prediction error increases, the step size doubles; if it decreases, the step size is reduced.

    • Standard Rates: Commercial voice quality is often achieved at 32Kbps32\,Kbps (using 44 bits per sample at 8kHz8\,kHz), but can go as low as 1616 or 24Kbps24\,Kbps.

    • ITU Recommendation G.728: Covers 16/24/32/46Kbps16/24/32/46-Kbps ADPCM and acts as a "common language" between European and North American PCM formats.

    • ITU Recommendation G.722: Codes 7.1kHz7.1\,kHz audio into 64Kbps64\,Kbps for high-quality ISDN speech.

    • Applications: Submarine systems, PABX leased lines (doubling capacity), and DECT cordless phones.

    • Limitation: Does not support modems or faxes above 4,800bps4,800\,bps.

GSM Speech Coding

  • Motivation: Cellular networks require efficient coding to maximize radio frequency usage.

  • Implementation:

    • A hybrid principle restricting processing delay to 65ms\le 65\,ms.

    • Original data rate was 13Kbps13\,Kbps, reduced to 7Kbps7\,Kbps in 1995.

    • The voice signal is divided into 20ms20\,ms slices.

    • An analysis filter identifies and subtracts the periodical high-power component, which is sent as parameters.

    • The remaining low-level error signal is waveform coded using APCM.

    • A synthesis filter at the receiver reconstructs the speech using the transmitted coefficients.

Quality Measurement and Network Constraints

  • Mean Opinion Score (MOS): A measurement based on human opinions of quality ranging from 1 (Bad) to 5 (Excellent).

    • G.711 PCM is typically rated as Excellent.

    • Vocoders often sound synthetic and represent a lower MOS.

  • Path Delay Constraints:

    • Round-trip delay should be less than 150ms150\,ms for high quality.

    • Deterioration is noticeable once delay exceeds 180ms180\,ms.

    • Geosynchronous satellites have a one-way delay of approximately 250ms250\,ms.

    • Echo is noticeable if delay exceeds 3030 to 50ms50\,ms.

  • Echo Cancellers: Required for long delays (satellites or GSM coding) or nonideal return loss in 4W/2W hybrids.

Power Levels and Decibels (dB)

  • Definition of Decibel (dB): A logarithmic measure of power change.

    • Gain formula: gdB=10log10(PoutPin)g_{dB} = 10 \log_{10}(\frac{P_{out}}{P_{in}})

    • Loss is represented as negative gain (e.g., an absolute loss of 1/101/10 is 10dB-10\,dB).

    • Gains in a chain are added (e.g., a gain of 100,000,000100,000,000 is 80dB80\,dB).

  • Voltage Gain Formula: If impedances differ:     gdB=20log10(VoutVin)+10log10(ZinZout)g_{dB} = 20 \log_{10}(\frac{V_{out}}{V_{in}}) + 10 \log_{10}(\frac{Z_{in}}{Z_{out}})

  • Absolute Power Level (dBm): Power relative to 1mW1\,mW.

    • PdBm=10log10(P1mW)P_{dBm} = 10 \log_{10}(\frac{P}{1\,mW})

    • Output power level calculation: Pout,dBm=Pin,dBm+gdBP_{out,dBm} = P_{in,dBm} + g_{dB}.

  • Engineering Example (Radio Relay):

    • P1=1W+30dBmP_1 = 1\,W \rightarrow +30\,dBm (since 10log10(1000/1)=3010 \log_{10}(1000/1) = 30).

    • P2=+30dBm+30dB110dB+30dB=20dBmP_2 = +30\,dBm + 30\,dB - 110\,dB + 30\,dB = -20\,dBm.

    • 20dBm-20\,dBm converts back to 10μW10\, \mu W.

The Digital Milliwatt

  • Purpose: A reference signal for analog levels in the network to ensure compatibility across international digital connections.

  • Definition: A specific sequence of 8 code words that, when decoded, produces a 1kHz1\,kHz sine wave at 0dBm0\,dBm power level.

  • dBm0: The measured signal power level compared to the reference level generated by the digital milliwatt.

  • Reference Rules:

    • Standard overload threshold for PCM coder is +3.14dBm0+3.14\,dBm0.

    • Average speech level in an analog channel is approximately 15dBm-15\,dBm.

    • European PCM digital milliwatt requires inverting bits 2, 4, 6, and 8 of the 8-word sequence.

    • Calibration involves adjusting internal amplifiers/attenuators so that 0dBm0\,dBm input produces 0dBm0\,dBm output at the decoder.