Speech Synthesis

1
Speech Synthesis

• User friendly machine must have complete voice
  communication abilities
• Voice communication involves
• Speech synthesis
• Speech recognition

2
Elements of Speech

• Some popular electronic speech synthesis are
  modeled directly from the human tract
• Other technique by putting together various
  sounds that are required to produce speech
• Therefore, it is important to learn the sound
  characteristics

3
Elements of Speech (cont)
4
Elements of Speech (cont)

• The vocal system can be broken into
• Lungs
• Larynx
• Vocal cavity
• Lungs provide power to system by forcing air up
  through the larynx and into the vocal cavity
• Vocal cords which made up of skin layers create
  sound when it flap or vibrate when air passes
  through
• The vibrating action generates several resonant
  frequencies within the vocal cavity (which has
  several harmonic frequency)

5
Elements of Speech (cont)

• Different sounds is created by changing the shape
  of the vocal cavity with throat, tongue, teeth
  and lips.
• Engineers use frequency analyzer called sound
  spectrograph to study speech.
• Spectrograph have shown that the range of most
  human speech is from 150 to 3600Hz. This
  represent a frequency bandwidth of 3450Hz.
• Bass singer or soprano - bandwidth of 15kHz (from
  10Hz to 15kHz)
• Volume ratio for human speech is about 16,000 to
  1, from shout to whisper

6
Elements of Speech (cont)

• However, we do not need to design a speech
  synthesizer which generate frequency from 10Hz to
  15kHz and volume ratio of 16,000 to 1
• Therefore there is a need to produce an
  intelligible speech. E.g Telephone system has
  a bandwidth of 3000Hz (from 300 to 3300 Hz) with
  10001 volume ratio.

7
Electronic Speech Synthesis (ESS)

• 2 techniques commonly used in ESS are
• Natural speech analysis/synthesis
• Artificial constrictive/synthesis

Natural speech analysis/synthesis

• Involves recording and subsequent playback of
  human speech
• Can be analog or digital
• Best choice to produce limited speech as in
  vending machine, appliances and automobiles.

8
Electronic Speech Synthesis (ESS) (cont)
Natural speech analysis/synthesis (cont)

• Involves analysis and synthesis
• Analysis phase Analyzed human speech, coded in
  digital form and stored
• Synthesis phase Recall digitized speech from
  memory and converted to analog to re-create the
  original speech waveform

9
Electronic Speech Synthesis (ESS) (cont)
Natural speech analysis/synthesis (cont)

• Digital analysis/synthesis method provides more
  flexibility than analog method since the stored
  words or phrases can be randomly accessed from
  computer memory.
• However, the vocabulary size is limited by the
  amount of memory available.
• For this reason, several different encoding
  techniques are used to analyzed and compress the
  speech waveform which attempt to discard
  unimportant part resulting in fewer bits being
  required.

10
Electronic Speech Synthesis (ESS) (cont)
Natural speech analysis/synthesis (cont)

• 2 types of digital analysis/synthesis
• Time domain analysis/synthesis
• Speech waveform is digitized in time domain
• Analog-gt digital (use ADC). The stored samples
  then passed through DAC to reproduce the speech.
• E.g. Telephone directory assistance
• Frequency domain analysis/synthesis
• Frequency spectrum of analog waveform is analyzed
  and coded
• Synthesis operation attempts to emulate human
  vocal tract electronically by using stored
  frequency parameters obtained from analysis

11
Electronic Speech Synthesis (ESS) (cont)
Artificial Constructive/Synthesis

• Created artificially by putting together the
  various sounds that are required to produce a
  given speech segment
• The most popular technique is called phoneme
  speech synthesis
• Phoneme The smallest phonetic unit in a language
  that is capable of conveying a distinction in
  meaning, as the m of mat and the b of bat in
  English
• Phonetic Representing the sounds of speech with
  a set of distinct symbols, each designating a
  single sound phonetic spelling
• E.g. f-ne-tik sim-bl phonetic symbol

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Electronic Speech Synthesis (ESS) (cont)
Artificial Constructive/Synthesis (cont)

• Allophone A predictable phonetic variant of a
  phoneme. For example, the aspirated t of top, the
  unaspirated t of stop, and the tt (pronounced as
  a flap) of batter are allophones of the English
  phoneme /t/.

13
Electronic Speech Synthesis (ESS) (cont)
Artificial Constructive/Synthesis (cont)

• Phoneme and allophones sounds are coded and
  stored in memory
• Software algorithm then connect the phonemes to
  produce a given word. Words strung together to
  produce phrase

14
Electronic Speech Synthesis (ESS) (cont)
Artificial Constructive/Synthesis (cont)

• There is also software that consists of a set of
  production rules that are used to translate
  written text into appropriate allophones code
  Text to speech
• With phoneme technique, a computer can produce
  unlimited vocabulary using minimum amount of
  memory

15
Electronic Speech Synthesis (ESS) (cont)
Summary of the various methods that are used for
electronic speech synthesis (ESS)
16
Time-Domain Analysis/Synthesis and Waveform
Digitization
17
Time-Domain Analysis/Synthesis and Waveform
Digitization (cont)

• Any method that converts the amplitude variations
  of speech to digital code for subsequent playback
  can be considered time-domain speech
  analysis/synthesis
• Time-domain speech analysis/synthesis involve 2
  operations
• Encoding human speech waveform to digitize and
  store the speech using analog to digital
  converter (ADC)
• Decoding the digitized speech into analog form
  for playback using digital to analog converter
  (DAC)

18
Time-Domain Analysis/Synthesis and Waveform
Digitization (cont)

• Low pass filter is connected to DAC output to
  smooth out the steps in the synthesized waveform
• Time-domain encoding attempt to reduce the amount
  of memory required to store digitized speech.
  Some examples are
• Simple Pulse-Code Modulation (PCM)
• Delta Modulation (DM)
• Differential Pulse-Code Modulation (DPCM)
• Adaptive Differential Pulse-Code Modulation
  (ADPCM)

19
Time-Domain Analysis/Synthesis and Waveform
Digitization (cont)
Simple Pulse-Code Modulation (PCM)

• Direct waveform digitization using ADC
• Things that control the quality of the digitized
  speech
• Sampling rate high sampling rate creates higher
  quality output
• ADC Resolution higher bit converter creates
  higher quality output
• To catch all subtleties of the waveform, it
  must be sampled about 30,000 times per second
• If each sample were converted to an 8 bit digital
  code, the data conversion rate would require
  8X30,000 or 240,000 bits of memory (data
  conversion rate 240,000 bit per second not
  practical)

20
Time-Domain Analysis/Synthesis and Waveform
Digitization (cont)
Simple Pulse-Code Modulation (SPCM) (cont)

• To reduce data rate, sampling rate must be
  reduced
• Experimentation has shown that acceptable speech
  can be created using a sampling rate of at least
  two times the highest frequency component in the
  speech waveform.
• Therefore 6000 (or 3000X2) conversion per second
  is the minimum sampling required to produce
  acceptable speech (most speech falls in the 300
  to 3000Hz).
• The ADC resolution determines the smallest analog
  increments that will be detected by the system.
  Acceptable speech can be synthesized using an
  8-bit ADC. (8 X 6,000 48,000 bps data rate).
  Therefore 10 second of speech needs 60,000 bytes
  of memory or 58.6kByte roughly

21
Time-Domain Analysis/Synthesis and Waveform
Digitization (cont)
22
Time-Domain Analysis/Synthesis and Waveform
Digitization (cont)
Simple Pulse-Code Modulation (SPCM) (cont)

• Problem Requires too much memory to produce
  acceptable speech for any length of time.
• The answer is to use
• Delta Modulation (DM)
• Differential Pulse Code Modulation (DPCM)
• Adaptive Differential Pulse Code Modulation
  (ADPCM)

23
Time-Domain Analysis/Synthesis and Waveform
Digitization (cont)
Delta Modulation (DM)

• Only single bit is stored for each sample of
  speech waveform, rather than 8 or 12 bits.
• The ADC still converts a sample to 8 or 12 bit
  value but then it is compared to the last sample
  value.
• If the present value is greater than the last
  value, the computer stores a logic 1-bit value.
  If less logic 0 is stored. Therefore, a single
  bit is used to represent data
• An integrator is used on the circuit output to
  convert the serial bit stream to analog waveform

24
Time-Domain Analysis/Synthesis and Waveform
Digitization (cont)
Delta Modulation (DM) (cont)

• Disadvantages
• The sampling rate must be high to catch all the
  details of the speech signal. E.g. Typical
  sampling rate 32,000 samples per second
  translates to 32,000 bps data rate. Thus, 10
  seconds of speech would require 39kbytes of
  memory (40 data reduction from 8-bit SPCM)
• Compliance Error/Slope Overload
• Results when speech waveform change so rapidly
  for a given sampling rate. The resulting
  digitization does not truly represent the analog
  waveform and produces audible distortion in the
  output.
• Can be solved by increasing sampling rate
  (however, will result in an increased data rate
  and more memory)

25
Time-Domain Analysis/Synthesis and Waveform
Digitization (cont)
Delta Modulation (DM) (cont)
26
Time-Domain Analysis/Synthesis and Waveform
Digitization (cont)
Differential Pulse-Code Modulation (DPCM)

• Same as DM but several bits are used to represent
  the actual difference between two successive
  samples rather than a single bit.
• Since speech waveform contains many duplicate
  sound and pauses, the change in amplitude from
  one sample to the next is relatively small as
  compared to the actual amplitude of a sample. As
  a result, less bits are required to store the
  difference value than the absolute sample value
• Difference between two successive samples can be
  represented with 6 or 7-bit value (1 sign bit
  (represent the slope of the input waveform) 5 or
  6 bits for difference value)

27
Time-Domain Analysis/Synthesis and Waveform
Digitization (cont)
Differential Pulse-Code Modulation (DPCM) (cont)

• E.g. A 7-bit DPCM system using a sampling rate of
  6,000 samples per second would require 51kB of
  memory for 10 seconds of speech.
• 7 X 6,000 X 10s 420,000 bit
• 420,000bit /8
• 52,500 byte
• gt 52,500/1024 51.3kB
• Bipolar DAC is used for playback to convert the
  successive difference values to a continuous
  analog waveform
• Have the same problems as DM. Overcome by
  increasing sampling rate and bit rate.

28
Time-Domain Analysis/Synthesis and Waveform
Digitization (cont)
Differential Pulse-Code Modulation (DPCM) (cont)
29
Time-Domain Analysis/Synthesis and Waveform
Digitization (cont)
Adaptive Differential Pulse-Code Modulation
(ADPCM)

• Is a variation of DPCM that eliminates the slope
  overload-overload problems
• Only 3 or 4 bits are required to represent each
  sample
• Waveform is sampled at 6000 samples per second
  with 8 or 12 bit ADC. The computer then subtract
  current sample value from previous to get
  differential value as DPCM. However, the
  differential value is then adjusted to compensate
  for slope using quantization factor.
• The quantization factor adjusts the differential
  value dynamically, according to the rate of
  change or slope of the input waveform. The
  adjusted differential value can then be
  represented using only 3 or 4 bits.

30
Time-Domain Analysis/Synthesis and Waveform
Digitization (cont)
Adaptive Differential Pulse-Code Modulation
(ADPCM) (cont)

• In addition to requiring fewer bit, the sampling
  rate of ADPCM can be reduced (to 4000 samples),
  since slope overload is minimized
• E.g. 4000 samples per second with 3 bit code
  results in data rate of 12,000 bps (i.e. 3 X
  4000). Therefore 10s of speech require 15kB
  memory. (I.e. (12,000/8) X 10s 15k
• However, 8000 samples per second and 4-bit code
  are more common gt 39kByte of memory for 10s of
  speech
• Disadvantage Needs sophisticated software