Speech coding techniques
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This document discusses speech coding techniques. It introduces speech processing which includes compression, manipulation, storage, transfer and reconstruction of speech. It explains that speech coding is needed for applications like mobile communication, voice over internet protocol, satellite broadcasting and PSTN networks. The document outlines various attributes of speech coders like low bit-rate, high speech quality and low computational complexity. It also describes different types of speech coders like waveform coders, source coders, time domain coders and frequency domain coders. Objective and subjective measures for evaluating performance of speech coders are discussed.
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Speech coding techniques are used to represent human speech in a digital form for applications like mobile communication and voice over IP. The main components of a speech coding system are speech encoding and decoding. Various coding techniques are used including waveform coding techniques like PCM and ADPCM, and source coding techniques like linear predictive coding (LPC) and vocoding. The aim is to enhance speech quality at a particular bitrate or minimize the bitrate at a given quality level, while considering factors like computational complexity, coding delay, and robustness to different speakers.
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3. Adaptive equalizers automatically adapt to changing channel properties over time using algorithms like LMS and RLS to update equalizer coefficients.
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Modulation involves adding information to a carrier signal. Digital modulation provides more information capacity, compatibility with digital services, higher security, better quality, and faster availability compared to analog modulation. Common digital modulation techniques include amplitude-shift keying (ASK), frequency-shift keying (FSK), phase-shift keying (PSK) and their variants. PSK techniques include binary PSK (BPSK), quadrature PSK (QPSK) and differential PSK (DPSK). QPSK transmits twice as much data as BPSK within the same bandwidth. DPSK avoids the need for a coherent reference signal at the receiver. Key considerations in modulation include power efficiency, bandwidth efficiency and bit error rate.
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2) The mean square error criterion aims to minimize the error between the received and desired signals when filtered by the equalizer. This can be solved using least squares or adaptive algorithms like LMS.
3) The least mean square algorithm approximates the steepest descent method to iteratively and adaptively update the equalizer filter taps to minimize the mean square error based only on instantaneous measurements. This makes it suitable for time-varying channels.
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Frequency Division Multiple Access (FDMA) is a channel access method where the available bandwidth is divided into multiple non-overlapping frequency bands and each user is assigned a specific frequency band. Each user can transmit or receive independently in its assigned frequency band without interference from other users. FDMA requires expensive bandpass filters for each frequency band and has strict linearity requirements for the transmission medium. The number of channels in an FDMA system is calculated by dividing the total available bandwidth minus the guard bands by the bandwidth of each individual channel.
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Thermal noise in communication systems is described by a zero-mean white Gaussian random process with a flat power spectral density (PSD) over all frequencies, making it white noise. The average power of white noise is infinite, so a solution is needed. The optimal receive filter, known as a matched filter, can be used to maximize the signal-to-noise ratio. Intersymbol interference (ISI) occurs when pulses interact with each other, such as through convolution with a low-pass channel filter, reducing performance if not addressed.
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The document discusses different techniques for multiplexing, which is the sharing of a transmission medium by multiple signals. It describes frequency division multiplexing (FDM), time division multiplexing (TDM) including synchronous and statistical TDM, wavelength division multiplexing (WDM), and code division multiplexing (CDM). TDM techniques like T-1 and ISDN use synchronous multiplexing to transmit multiple digital signals over a single circuit simultaneously.
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The document discusses speech processing and vocoding. It begins by defining speech and how it is produced, including voiced and unvoiced sounds. It then describes the human speech production system and various speech coding techniques like waveform coding, vocoding, and analysis-by-synthesis coding. Finally, it provides details on the G.729 speech codec, including its operations, process flow, specifications, and how it achieves speech compression to 8 kbps from the original 128 kbps.
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Quantization is the process of mapping continuous range of values to a finite set of values. It involves rounding samples to the nearest quantization level, changing infinite precision values to finite precision. For a given input signal sampled at 8 samples per second ranging from -1 to 1, quantization with 2 bits would result in 4 quantization levels spaced 0.5 units apart. The quantized values and errors can be calculated, with the errors assumed to be uniformly distributed between -0.25 and 0.25.
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- 1. Speech Coding Techniques BY: Kailash Karki 152040006 KL University Andhra Pradesh,India Kailash202012@gmail.com
- 2. Introduction Speech processing includes: o Compression o Manipulation o Storage o Transfer o Output(Reconstruction) Can be considered as a special case of digital signal processing.
- 3. Why speech coding is needed?
- 5. Attributes of Speech Coders • Low bit-rate • High speech quality • Channel errors • Low memory requirements • Less computational complexity • Low coding delay
- 6. Hierarchy of Speech Coders Speech Coders Waveform Coders Source Coders Time Domain LPC VOCODERSFrequency Domain NonDifferential Differential Delta ADPCM SBC ATCPCM
- 7. Performance Evaluation of Speech Coders • Objective Measures • Subjective Measures Quality Scale Score Listening Effort Scale Excellent 5 No effort required Good 4 No appreciable effort required Fair 3 Moderate effort required Poor 2 Considerable effort required Bad 1 No meaning understood with reasonable effort
- 8. Performance of Coders Coder MOS 64 kbps PCM 4.3 14.4 kbps QCELP13 4.2 32 kbps ADPCM 4.1 8 kbps ITU-CELP 3.9 8 kbps CELP 3.7 13 kbps GSM Codec 3.54 9.6 kbps QCELP 3.45 4.8 kbps CELP 3.0 2.4 kbps LPC 2.5
- 9. Applications: • Mobile Communication • Voice over internet protocol • Satellite broadcasting • PSTN Network etc.