SPEECH PROCESSING

ASSIGNMENT – 2
(Narasimha,VU21EECE0100447)

Q) Explain voice response block diagram

A) 1. Block Diagram of a Voice Response System

Components:

• Message Requests:

o This module accepts requests in the form of messages from users or systems.
• Message Composition Program:

o This component processes the message requests and determines the appropriate
response.

• Message Formation Rules:

o These rules govern how messages are structured and formatted before being sent out
as responses.

• Vocabulary Storage:

o This is a database that contains all possible responses or vocabulary that can be used in
the output messages.

• Output Message:
 o This is the final message produced by the system, which is delivered to the user based
on the input requests.

2) Block Diagram of an All-Digital Voice Response System

Components:

• Speech Input:

o The initial stage where analog speech is input into the system.
• Analog-to-Digital Coder:

o Converts the analog speech input into a digital format for processing. This is essential
for further digital manipulation.

• Vocabulary Preparation System:

o Prepares the vocabulary for use in the system, ensuring that responses are readily
available in the digital format.

• Digital Storage:

o This module stores the digital representations of the vocabulary and any pre-recorded
responses.

• Message Composition System:

o Similar to the first system, this component composes messages based on requests and
utilizes stored vocabulary for response formulation.

• Digital-to-Analog Decoder:

o Converts the digital response back into an analog signal for output, making it
understandable in a human-readable form.
3) Multiline Digital Voice Response System

The block diagram you shared represents a Multiline Digital Voice Response System, which processes
multiple message requests (MR) and generates corresponding output messages (OM). Here's a detailed
breakdown:

Key Components:

1. Natural Speech Input:

o The system starts with natural speech as the input, which is then processed through
various stages to generate appropriate responses.

2. ADPCM Coder (Adaptive Differential Pulse Code Modulation Coder):

o This coder converts the analog speech input into a compressed digital format using
ADPCM coding. ADPCM is used to efficiently encode speech signals while
maintaining a good balance between quality and data compression.

3. Vocabulary Preparation Program:

o This module is responsible for preparing and managing the system's vocabulary. It
ensures that the system has a set of predefined words or phrases available for response
generation.

o The system likely uses digital recordings of vocabulary, prepared in advance for various
scenarios.

4. Vocabulary Storage:

o The vocabulary, once prepared, is stored in the Digital Storage, often using a fixed
head disk for fast access. This storage contains pre-recorded and digitally stored speech
segments.
 5. Message Composition Program:

o This is the core processing unit. It takes in multiple Message Requests (MR), labeled
from MR₁ to MR₁₀, meaning the system can handle up to 10 message requests
concurrently.

o Based on the requests, the program assembles appropriate responses by fetching the
required vocabulary from the storage.

6. ADPCM Decoders:

o After the message composition, each output message is sent to its respective ADPCM
Decoder (e.g., ADPCM Decoder 1, Decoder 2, ... Decoder 10).

o These decoders decompress the encoded digital data back into a suitable format for
speech output. Each decoder corresponds to a different output message line.

7. Output Messages (OM₁ to OM₁₀):

o The decoders send the output to 10 different channels (OM₁ to OM₁₀), each providing
an audio message based on the input request.

o This allows the system to respond to multiple requests in parallel, a crucial feature in
systems handling high volumes of queries

4) voice response system based on formant representation

The block diagram in Figure 9.5 illustrates a voice response system based on formant
representation. This system analyzes spoken words and synthesizes speech using key speech
characteristics like formants, pitch, and amplitude.

Key Components:

1. Spoken Single Words:

o The system begins by receiving single spoken words as input. These words are
analyzed for key speech characteristics.

2. Analog to Digital Conversion:

 o The spoken words are first converted from an analog form to digital signals for further
processing by the system.

3. Formant Analysis Program:

o This module extracts the formants (F₁, F₂, F₃), which are the resonant frequencies of
the vocal tract that are crucial for distinguishing different speech sounds. These
frequencies are essential for synthesizing realistic-sounding speech.

4. Pitch Analysis Program:

o This program analyzes the pitch (P) of the spoken words, which refers to the
fundamental frequency of the speech. Pitch provides information about the tone and
intonation of the speech, crucial for natural-sounding voice responses.

5. Amplitude Analysis Program:

o The amplitude (Aₑ, Aₙ) of the speech is analyzed in this stage. Amplitude measures the
loudness or intensity of the sound, which helps in accurately reproducing the volume
levels of the speech during synthesis.

6. Fricative Pole-Zero Analysis Program:

o This module analyzes the fricative sounds of speech, such as “s” or “f,” which are
characterized by noise-like energy. It identifies the pole-zero characteristics (Fₚ, Fₓ) to
capture the specific sound qualities of fricative phonemes.

7. Data Storage for Formant-Coded Words:

o The results of these analyses are stored in the formant-coded word library, where
digital representations of words are kept for later retrieval and synthesis. This is
effectively the system’s vocabulary.

8. Answer-Back Program:

o The system can respond based on typed alphabetic characters (text input), where the
corresponding word sequence is demanded by the program.

9. Program for Concatenating Words:

o The stored digital speech components (words) are concatenated to form complete
phrases or sentences, guided by predefined rules. This ensures that the speech output
flows naturally.

10. Timing and Pitch Data:

• Timing data and pitch variation can be either calculated internally or supplied externally to
help regulate how quickly the system speaks and with what intonation. This data helps in
creating natural-sounding speech synthesis with proper rhythm and emphasis.

11. Hardware Digital Speech Synthesizer:

• This hardware module generates speech from the concatenated digital words, controlling
formant, pitch, and amplitude values. It forms the core of the speech synthesis process.

12. Digital to Analog Conversion:

• The synthesized digital speech is converted back to analog form using a digital-to-analog
converter, allowing the speech to be played through a speaker as audible sound.
2) Explain speaker verification
A) Speaker Verification Theory:

Speaker verification refers to the process of confirming a speaker's identity based on their voice
characteristics. Unlike speaker identification, which determines "who" the speaker is from a known
set of speakers, speaker verification answers the question, "Is this person who they claim to be?" It's a
crucial element in systems like biometric authentication, secure communications, and voice-activated
access control.

Decision Rule for Speaker Verification:

In the decision rule illustrated in the image, a simple statistical approach is applied to verify or reject a
speaker, using probabilistic measures.

The decision rule is as follows:

• Verify speaker if:

pi(x)>ci⋅pav(x)

• Reject speaker if:

pi(x)<ci⋅pav(x)

Where:

• pi(x)p_i(x)pi(x) represents the probability distribution of the measured voice characteristics

xxx for speaker iii, which reflects how likely the measured voice matches the claimed speaker's
voice profile.

• pav(x)p_{\text{av}}(x)pav(x) represents the average probability distribution over all

speakers in the system (ensemble). This average distribution serves as a baseline for general
voice characteristics.

• cic_ici is a constant specific to speaker iii, which helps in determining the error balance for that
speaker. Adjusting this constant allows for controlling the trade-off between false acceptances
(accepting an impostor) and false rejections (rejecting the correct speaker).

Explanation:

• The decision rule compares the likelihood of the current speaker's voice sample pi(x)p_i(x)pi
(x) to the average voice sample pav(x)p_{\text{av}}(x)pav(x), weighted by a constant cic_ici.
If the speaker's voice sample exceeds this weighted threshold, they are verified as the claimed
speaker. If not, the system rejects their claim.

• The constant cic_ici is crucial for balancing the two types of errors in speaker verification:

1. False Rejection Rate (FRR): Occurs when the system incorrectly rejects the correct
speaker. A smaller cic_ici can lower FRR by making it easier for the speaker to be
verified.

2. False Acceptance Rate (FAR): Happens when the system incorrectly verifies an
impostor as the speaker. A larger cic_ici can reduce FAR by making verification stricter.

By carefully tuning cic_ici, the verification system can control the relative importance of minimizing
these errors, depending on the application requirements.
Types of Errors in Speaker Verification:

1. False Acceptance (Type I Error): The system incorrectly verifies an impostor as the legitimate
speaker. It occurs when the probability distribution of the impostor's voice closely matches the
legitimate speaker's distribution.

2. False Rejection (Type II Error): The system fails to verify the legitimate speaker, rejecting
them despite their correct identity. This can happen when the speaker's voice characteristics
deviate from their previously enrolled model due to noise, illness, or other factors.

Applications of Speaker Verification:

• Biometric Authentication: Speaker verification is used in securing access to devices, systems,

or facilities. It ensures that only authorized individuals, verified by their unique voice patterns,
can gain access.

• Banking and Finance: Some banks use voice verification as an added layer of security for
transactions or customer service interactions.

• Telecommunications: Many voice-controlled systems, such as phone-based customer service

systems, use speaker verification to verify the caller's identity before sharing sensitive
information.

Techniques for Speaker Verification:

1. Template Matching: A user's voice is compared to a stored template, and verification is

performed based on similarity.

2. Gaussian Mixture Models (GMMs): Probabilistic models like GMMs are used to represent
voice characteristics. The likelihood of a match is calculated based on how well the new voice
data fits the speaker's model.

3. Deep Learning Approaches: Neural networks are increasingly used to model voice
characteristics for speaker verification, especially in complex and noisy environments.

By adjusting the parameters of the decision rule, including cic_ici, speaker verification systems can be
fine-tuned for different levels of security, balancing the trade-off between security (minimizing false
acceptances) and user convenience (minimizing false rejections).
3) Explain speaker recogintion
A) The diagram in the image represents the general structure of a speaker
recognition system.

Let's break down each component in the system:

1. Signal Processor

• Input: A speech signal s(n)s(n)s(n) from the speaker.

• Function: The signal processor extracts relevant features from the raw speech signal. This could
include techniques like Fourier transforms, Mel-frequency cepstral coefficients (MFCCs),
or other feature extraction methods that reduce the signal into a form suitable for comparison.

• Output: A pattern vector xxx, which is a mathematical representation of the speaker’s voice
characteristics.

2. Reference Patterns

• These are pre-recorded voice templates of known speakers, which are stored in the system.
Each reference pattern represents the voice features of a particular speaker, which will be used
for comparison during the recognition process.

3. Comparison (Distance Measurement)

• Input: The pattern vector xxx from the signal processor and the reference patterns of known
speakers.

• Function: The system compares the extracted features of the input speech (pattern vector) with
the stored reference patterns of speakers. It computes a distance measure, which quantifies the
similarity between the input voice and each reference pattern.

• Output: A distance score DDD, indicating how close the input voice matches each reference
pattern. The smaller the distance, the more similar the voices are.

4. Decision Logic

• Input: The distance DDD from the comparison module.

• Function: Based on the computed distance, the decision logic determines whether the input
speech matches any of the known speakers or if it belongs to an unknown individual. This step
 could involve a threshold decision rule: if the distance is below a certain threshold, the speaker
is considered identified; otherwise, they are rejected.

• Output: Identification of the speaker, which is the final decision about the speaker's identity.

Summary:

• The signal processor transforms the input speech into a pattern vector.

• This vector is compared to reference patterns of known speakers by calculating the distance
between them.

• The decision logic interprets the distance and decides if the speaker is identified or not.