En 542839Microchip DSC Audio and Speech Solutions

1 2007 Microchip Technology Incorporated. All Rights Reserved. DSC Audio and Speech Solutions Web Seminar Slide 1

Microchip DSC Audio and Speech Solutions

Welcome to this seminar on Microchips Digital Signal Controller Audio and Speech Solutions. My name is Sunil Fernandes. I am an Applications Engineer with the Digital Signal Controllers group here at Microchip Technology.


Agenda

Overview Speech Coding Solutions Speech Processing Solutions Audio Solutions Summary

This seminar is intended to provide an introduction to Microchips Speech Coding and Processing Solutions. We will start with an overview of the Speech Coding and Processing Solutions that Microchip offers. We will then discuss some features of every solution. Some of the working principles of these algorithms will be discussed. We will then talk about the latest addition to Microchips Audio Solutions group , The Audio PICTail Plus board. We will conclude with a summary of this presentation and some useful references.


Speech Audio Solutions Overview

The Speech and Audio Solutions can be classified into three groups.


Speech Audio Solutions OverviewSOLUTIONS

SPEECH CODING SPEECH PROCESSING HARDWARE

The Speech Coding algorithms are used for purposes of changing the representation of speech signals with an objective to save memory requirements and bandwidth. These software libraries implement compression using waveform encoding and parametric encoding. The Speech processing libraries on the other hand are useful for processing the digitized speech signal. These software libraries eliminate hardware burden, improve audible speech quality and can be used to leverage the applications user interface. The Speech and Audio hardware represents hardware boards which are useful for demonstrating and testing speech and audio applications. In this seminar, we will specifically talk about the Audio PICTail Plus board.



WAVEFORMCODERS

VOCODERS

SOLUTIONS


The Speech coding libraries can be classified into waveform coders and vocoders. Waveform coders exploit the redundancies in the signal waveform to decrease the number of bits used to represent a speech frame.



WAVEFORMCODERS

VOCODERS

G.711 G.726A

SOLUTIONS


G.711 and G.726A are examples of waveform coders.



WAVEFORMCODERS

VOCODERS

G.711 G.726A

SPEEX

SOLUTIONS


Vocoders encode a speech frame as a set of parameters. These parameters are then used by the decoder to re-generate the corresponding speech signal. The SPEEX algorithm is an example of a vocoder. Typically vocoders are capable of better compression than waveform coders, but they require more computational resources.



Noise Suppression

WAVEFORMCODERS

VOCODERS

G.711 G.726A

SPEEX

SOLUTIONS


The Speech Processing algorithms comprise of signal processing algorithms that perform different functions on speech signals. The Noise Suppression algorithm reduces the background noise in a speech signal.



Noise Suppression

Acoustic Echo Cancellation

WAVEFORMCODERS

VOCODERS

G.711 G.726A

SPEEX

SOLUTIONS


The Acoustic Echo Cancellation reduces the echo caused due to acoustic coupling between a speaker and microphone.

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2007 Microchip Technology Incorporated. All Rights Reserved. DSC Audio and Speech Solutions Web Seminar Slide 10


Noise Suppression


Line Echo Cancellation

WAVEFORMCODERS

VOCODERS

G.711 G.726A

SPEEX

SOLUTIONS


Line Echo cancellation reduces the echo caused due to impedance mismatch in telecommunication networks.

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Noise Suppression



Speech Recognition

WAVEFORMCODERS

VOCODERS

G.711 G.726A

SPEEX

SOLUTIONS


The Speech Recognition library provides a speaker independent speech recognition system which can be used to implement a speech based command interface.

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Noise Suppression



Speech Recognition

WAVEFORMCODERS

VOCODERS

G.711 G.726A

SPEEX

SOLUTIONS


Audio PICTail Plus

Microchip offers a number of hardware development boards for testing and developing speech applications. In this seminar, we will talk about the latest addition to the Audio Solutions family, The Audio PICTail Plus Board. This board demonstrates a low cost audio playback technique. Lets start by discussing the Speech Coding solutions.

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G.711

The G.711 algorithm implements the ITU-T G.711 coding and decoding standard for speech signals.

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G.711 Uses a logarithmic curve based compression.

This coding mechanism is based on a logarithmic curve which allocates smaller quantization steps for small amplitude signals and larger quantization steps for large amplitude signals. The idea is to use less bits to encode large amplitude and more bits to encode small amplitude.

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G.711 Uses a logarithmic curve based compression. Waveform coder.

G.711 is a waveform coder since it encodes the amplitude of the signal.

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G.711 Uses a logarithmic curve based compression. Waveform coder. A-law and U-law

Depending on the geographic location, two logarithmic curves are used. These are referred to as A-law and u-law. Europe uses A-law whereas the United States uses U-law.

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G.711 Uses a logarithmic curve based compression. Waveform coder. A-law and U-law 2:1 compression ratio

The G.711 coding method provides a compression ratio of 2:1

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G.711 Uses a logarithmic curve based compression. Waveform coder. A-law and U-law 2:1 compression ratio Typical bit rates of 64kbps

which results in typical bit rates of 64Kbps.

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G.711 Uses a logarithmic curve based compression. Waveform coder. A-law and U-law 2:1 compression ratio Typical bit rates of 64kbps Used widely in legacy toll quality voice circuits.

The G.711 coding standard is used most widely in toll quality voice circuits.

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G.711 Uses a logarithmic curve based compression. Waveform coder. A-law and U-law 2:1 compression ratio Typical bit rates of 64kbps Used widely in legacy toll quality voice circuits. MOS rating of 4 - 4.5 out of 5.

The MOS or the Mean Opinion Score of a coding method is a subjective measure of the quality of the coder. There is always loss of some data when a signal is encoded and decoded. The MOS provides a subjective mechanism for specifying the impact of the coding and decoding process on the quality of the speech signal. The G.711 coder has a high MOS rating since the compression ratio is only 2:1 and the typical data rate is high.

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G.711 Uses a logarithmic curve based compression. Waveform coder. A-law and U-law 2:1 compression ratio Typical bit rates of 64kbps Used widely in legacy toll quality voice circuits. MOS rating of 4 - 4.5 out of 5. Does not require DSP.

The G.711 algorithm does not require any DSP instructions and so is available for both, dsPIC and PIC24H devices. It is not computationally intensive and usually is implemented using a table look up or a bit manipulation scheme.

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G.711 Uses a logarithmic curve based compression. Waveform coder. A-law and U-law 2:1 compression ratio Typical bit rates of 64kbps Used widely in legacy toll quality voice circuits. MOS rating of 4 - 4.5 out of 5. Does not require DSP. 1 MIPS, 3.5KB Flash and 3.5KB RAM

It requires 1 MIPS of processing power, 3.5KB program flash memory and 3.5KB of data RAM.

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G.711 Uses a logarithmic curve based compression. Waveform coder. A-law and U-law 2:1 compression ratio Typical bit rates of 64kbps Used widely in legacy toll quality voice circuits. MOS rating of 4 - 4.5 out of 5. Does not require DSP. 1 MIPS, 3.5KB Flash and 3.5KB RAM Its free and comes with the source code.

The library is available for free and comes with the source code.

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G.726A Uses feedback ADPCM based compression

The G.726A Speech coder implements the ITU-T G.726A recommendation and uses Adaptive Differential Pulse Code Modulation technique to minimize the number of bits required to encode the signal. Differential pulse code modulation encodes the difference between the current sample and a linear combination of certain number of past input samples. Statistically, this difference would be small and would require less bits to encode.

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G.726A Uses feedback ADPCM based compression Waveform coder

The G.726A algorithm is another example of a waveform coder as it operates on the speech waveform amplitude.

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G.726A Uses feedback ADPCM based compression Waveform coder 3.2:1, 4:1, 5.33:1 or 8:1 compression ratio

The algorithm supports different bit rates. Depending on the chosen rate, compression rates of 3.2:1, 4:1, 5.33:1 and 8:1 are possible.

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G.726A Uses feedback ADPCM based compression Waveform coder 3.2:1, 4:1, 5.33:1 or 8:1 compression ratio 40/32/24/16 kbps

This translates to bit rates of 40/32/24 and 16 kbps.

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G.726A Uses feedback ADPCM based compression Waveform coder 3.2:1, 4:1, 5.33:1 or 8:1 compression ratio 40/32/24/16 kbps Used typically in record/playback and

communication applications.

The G.726A coder is typically used in record / playback type of applications and in digital communication applications such as walkie-talkies. The library is available for dsPIC30F and dsPIC33F devices. It requires 15 MIPS, 6KB of program flash and 4KB of data RAM.

.

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communication applications. 15 MIPS, 6KB Flash and 4KB RAM

The library is available for dsPIC30F and dsPIC33F devices. It requires 15 MIPS, 6KB of program flash and 4KB of data RAM.

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communication applications. 15 MIPS, 6KB Flash and 4KB RAM MOS of 3.5 - 4.5 out of 5.0

The G.726A has a Mean Opinion Score between 3.5 and 4.5 depending on the selected bit rate.

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communication applications. 15 MIPS, 6KB Flash and 4KB RAM MOS of 3.5 - 4.5 out of 5.0 Evaluation version available for download

An evaluation version of the library is available. You can contact your local Microchip Sales office for more details.

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SPEEX Uses CELP based compression techniques

The SPEEX Speech compression library uses Code Excited Linear Prediction technique to achieve a high compression ratio.

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SPEEX Uses CELP based compression techniques Vocoder parametric coder.

SPEEX is an example of a vocoder. It operates by obtaining parameters from a speech data frame. These parameters are then used to re-produce the speech data frame in the decoder.

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SPEEX Uses CELP based compression techniques Vocoder parametric coder. Based on an open-source algorithm

The SPEEX algorithm is an open source project. More details of the compression technique and the algorithm are available at www.speex.org

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SPEEX Uses CELP based compression techniques Vocoder parametric coder. Based on an open-source algorithm Compression ration of 16:1

The SPEEX coder features a compression ratio of 16:1.

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SPEEX Uses CELP based compression techniques Vocoder parametric coder. Based on an open-source algorithm Compression ration of 16:1 Bit rates of 8Kbps

For a standard bit rate of 128Kbps, this results in compressed bit rate of 8Kbps.

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SPEEX Uses CELP based compression techniques Vocoder parametric coder. Based on an open-source algorithm Compression ration of 16:1 Bit rates of 8Kbps Used in record / playback and communication

applications.

This makes the SPEEX algorithm an ideal candidate for record and playback applications, VoIP or other digital communication applications.

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applications. MOS of 3.7 4.2 out of 5.0

The SPEEX algorithm has a Mean Opinion Score of 3.7 to 4.2.

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applications. MOS of 3.7 4.2 out of 5.0 20 MIPS, 30KB Flash and 7KB RAM

Naturally, due to the high compression rate that it produces, the SPEEX coder is DSP intensive and requires 20 MIPS for execution. It requires 30KB of program flash and 7KB of Data RAM for its operation. The library is available for both the dsPIC30F and the dsPIC33F devices.

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applications. MOS of 3.7 4.2 out of 5.0 20 MIPS, 30KB Flash and 7KB RAM Evaluation version available for download

An evaluation version of the library is available. You can contact your local Microchip Sales office for more details.

We will now discuss the Speech Processing Libraries.

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Noise Suppression Suppresses noise interference in a speech

signal Does not require reference noise input Provides 10-20 dB of reduction Library optimized for voice band applications Useful for intercoms, speaker phones and

headsets Includes sampling rate conversion routines.

The Noise Suppression library suppresses background and ambient noise in a speech signal. It does this by tracking and attenuating frequency components that contribute to the noise in a speech segment.

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signal Does not require reference noise input

Unlike other algorithms that require a reference noise input to specifically track the background noise, this algorithm uses periods of no speech to update its noise estimator. Hence its does not require a reference noise input which makes the algorithm easier to use.

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signal Does not require reference noise input Provides 10-20 dB of reduction

The algorithm achieves noise suppression in the range of 10-20dB.

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signal Does not require reference noise input Provides 10-20 dB of reduction Library optimized for voice band applications

The library is optimized for operation in the 300 to 3300Hz frequency range which corresponds to voice band frequencies.

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headsets

This makes the Noise Suppression library ideal for use in applications such as intercoms, speaker phones, headsets and hands free kits.

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headsets Includes sampling rate conversion routines.

The library also includes sample rate conversion routines to down-sample and up-sample between sampling frequencies adjacent to 8000Hz.

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Acoustic Echo Cancellation Suppresses acoustic echo caused by speaker-

microphone coupling

The Acoustic Echo Cancellation library or AEC for short is based on an adaptive filter algorithm that suppresses echoes caused due to coupling between a speaker and the microphone. An example of where such an coupling could take place is in a speaker phone application where the microphone is in close proximity of the speaker.

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microphone coupling Compliant with ITU-T G.167 recommendation

The library meets the requirements specified by the ITU-T G.167 recommendation for Acoustic Echo Controllers.

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microphone coupling Compliant with ITU-T G.167 recommendation Provides 16,32 and 64 millisecond echo tail

length support

It supports echo tail lengths of 16, 32 and 64 milliseconds. This allows the AEC to be used in short echo tail length environments such as cars or long tail length environments such as conference rooms.

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length support Provides 40-50 dB of echo cancellation

Depending on the nature of the acoustic environment, the AEC algorithm provides echo cancellation in the range of 40-50 dB.

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length support Provides 40-50 dB of echo cancellation Optimized for voice band applications

The algorithm is optimized for operation in 300 to 3300Hz frequency range which corresponds to voice band frequencies.

52




length support Provides 40-50 dB of echo cancellation Optimized for voice band applications Useful for speaker phones and intercom

systems

This makes the AEC algorithm ideal for use in applications such as speaker phones, intercom systems and hands free kits.

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length support Provides 40-50 dB of echo cancellation Optimized for voice band applications Useful for speaker phones and intercom

systems Library includes sampling rate conversion

routines.

The library also includes sample rate conversion routines to down-sample and up-sample between sampling frequencies adjacent to 8000Hz.

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REMOTE IN

REMOTE OUT

NEAR IN

NEAR OUT

COMMUNICATION SYSTEM

Lets take a look at how an Acoustic Echo Cancellation system works. Consider an example of a communication system. This system could be a hands free kit or a speaker phone. The remote in signal is the output from the remote calling party. The remote out signal is the input to the remote calling party. Similarly at the near end, the near in signal is connected to a speaker-amplifier. The near out signal is obtained from a local microphone. The near end signals represent the user audio interface and are usually responsible for generating acoustic echo.

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REMOTE IN

REMOTE OUT

NEAR IN

NEAR OUT

COUPLING


The echo is caused by acoustic coupling between the speaker and the microphone. The coupling typically includes components due to the direct path between the speaker and the microphone and an indirect path due to room acoustics.

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REMOTE IN

REMOTE OUT

NEAR IN

NEAR OUT

COUPLING

ECHO


The remote calling party hears this signal as an echo of the transmitted speech.

The AEC system works by attempting to approximate the acoustic environment that causes the coupling.

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REMOTE IN

REMOTE OUT

NEAR IN

NEAR OUT

COUPLING

ECHOAdaptive Filter

-

+


An adaptive filter performs this function. The filter continuously updates its model of the acoustic environment. The output of the filter is therefore an approximation of the echo component contained in the near out signal. By subtracting this approximation from the near out signal, the AEC algorithm suppresses the echo contained in the remote out signal.

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Suppresses echo caused due to wire line conversion hybrids in telephone lines.

The Line Echo Cancellation library or LEC for short is based on an adaptive filter algorithm and suppresses echoes caused by impedance mismatch or improper termination in a communication line. An example of where such a coupling could take place is in a two wire to four wire conversion hybrid in telephone lines.

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Compliant with ITU-T G.168 recommendation.

The library meets the requirements specified by the ITU-T G.168 recommendation for Line Echo Controllers.

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Compliant with ITU-T G.168 recommendation. Useful for VoIP and telecommunication interface

circuits.

The LEC algorithm can be used for front end processing of speech data acquired from a telecommunication interface. Typical applications include Voice over IP and speaker phones.

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circuits. Configurable tail lengths of 16,32 and 64 msec.

It supports echo tail lengths of 16, 32 and 64 milliseconds. This allows the application to tailor the LEC library to suit the requirements of the communication system. The algorithm features a typical convergence rate of 30dB/sec and echo cancellation of up to 40dB.

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circuits. Configurable tail lengths of 16,32 and 64 msec. Optimized for voice band applications

The algorithm is optimized for operation in 300 to 3300Hz frequency range which corresponds to voice band frequencies . The library also includes sample rate conversion routines to down-sample and up-sample between sampling frequencies adjacent to 8000Hz.

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REMOTE OUT

REMOTE IN NEAR IN

NEAR OUT

NETWORK SYSTEM

Lets take a look at how an Line Echo Cancellation system works. Consider an example of a network system. This system could be a speaker phone. The remote in signal is the output from the remote calling party. The remote out signal is the input to the remote calling party. Similarly at the near end, the near in signal is connected to a speaker-amplifier. The near out signal is obtained from a local microphone.

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COUPLING

REMOTE OUT

REMOTE IN NEAR IN

NEAR OUT

NETWORK SYSTEM

Line echo is generated by a coupling between the Remote Out and Remote In signals.

The echo is caused by improper termination or an impedance mismatch in the remote circuit. The impedance mismatch causes the transmitted signal to be reflected back. This leads to a portion of the remote out signal getting coupled to the remote in signal.

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COUPLINGECHO

REMOTE OUT

REMOTE IN NEAR IN

NEAR OUT

NETWORK SYSTEM

The near end party hears this signal as an echo of the transmitted speech.

The LEC system works by attempting to approximate the electrical environment that causes the coupling.

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COUPLINGECHO

REMOTE OUT

REMOTE IN NEAR IN

NEAR OUT

NETWORK SYSTEM

Adaptive Filter

-+

An adaptive filter performs this function. The filter continuously updates its model of the electrical environment. The output of the filter is therefore an approximation of the line echo component contained in the remote in signal. By subtracting this approximation from the remote in signal, the LEC algorithm suppresses the echo contained in the remote in signal.

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Speech Recognition

The Speech Recognition library provides voice control for embedded applications.

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Speech Recognition

Speaker Independent recognition

The library is pre-trained by a demographic cross section of male and female US English speakers and is speaker independent. The algorithm uses the Hidden Markov Model Processing technique for identifying the spoken word.

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Speech Recognition

Speaker Independent recognition PC based application word library builder

A PC based word library builder program creates a custom library from a

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Speech Recognition

Speaker Independent recognition PC based application word library builder Up to 100 words vocabulary (American English)

master library of 100 common words.

This PC based application generates source code files containing the Hidden Markov Models for selected words.

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Speech Recognition

Speaker Independent recognition PC based application word library builder Up to 100 words vocabulary (American English) Supports multiple noise profiles

The models accommodate three types of noise profile - automobile, office and white noise.

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Speech Recognition

Speaker Independent recognition PC based application word library builder Up to 100 words vocabulary (American English) Supports multiple noise profiles Useful for any voice activated applications such

as home appliances.

The Speech recognition library is an ideal front end for hands-free products such as modern appliances, security panels and cell phones.

We will now take a look at the latest addition to Microchips Speech and Audio Hardware solutions, the Audio PICTail Plus Board.

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Audio PICTail Plus Board

The Audio PICTail Plus board works with the dsPIC33F and PIC24H devices. The board is a part of the PICTail Plus family and plugs into the PICTail slot on the Explorer 16 Development board.

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OUTPUTCOMPARE

(dsPIC/PIC24H)

DCI & I2C(dsPIC)

12-bit ADC(dsPIC/PIC24H)

SPI(dsPIC/PIC24H)

While using the dsPIC33F devices, the Audio PICTail Plus board uses the Output Compare module, DCI module, I2C module, 12-bit ADC and the SPI module. While using a PIC24H device, the Audio PICTail Plus board use the Output Compare module, 12-bit ADC and the SPI module.

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OUTPUTCOMPARE

(dsPIC/PIC24H)

DCI & I2C(dsPIC)


SPI(dsPIC/PIC24H)

PRE-AMPLIERLINE

MIC

The Audio PICTail Plus board accepts either a microphone or a line level audio signal.

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ANTI-ALIASINGFILTER

OUTPUTCOMPARE

(dsPIC/PIC24H)

DCI & I2C(dsPIC)


SPI(dsPIC/PIC24H)

PRE-AMPLIERLINE

MIC

The signal is amplified and then processed by an anti-aliasing filter. This filter has a corner frequency at 3300Hz. The output of the filter feeds into the 12-bit ADC on the dsPIC or PIC24H device.

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ANTI-ALIASINGFILTER

16-BITCODEC

OUTPUTCOMPARE

(dsPIC/PIC24H)

DCI & I2C(dsPIC)


SPI(dsPIC/PIC24H)

PRE-AMPLIERLINE

MIC

The output of the pre-amplifier also feeds into a 16 bit audio codec. A dsPIC device can access the codec using the DCI and I2C modules. The 16 bit codec allows the user to consider different design choices while designing the end application.

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LOW PASSFILTER

ANTI-ALIASINGFILTER

16-BITCODEC

OUTPUTCOMPARE

(dsPIC/PIC24H)

DCI & I2C(dsPIC)


SPI(dsPIC/PIC24H)

PRE-AMPLIERLINE

MIC

The Output Compare module on dsPIC or the PIC24H device generates a Pulse Width Modulated signal. The duty cycle of signal is modulated by an audio signal.

The Low Pass Filter de-modulates the PWM signal.

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LOW PASSFILTER

ANTI-ALIASINGFILTER

16-BITCODEC

OUTPUTCOMPARE

(dsPIC/PIC24H)

DCI & I2C(dsPIC)


SPI(dsPIC/PIC24H)

HEAD-PHONE/LINE

AMPLIFIER HEAD PHONES/LINE

PRE-AMPLIERLINE

MIC

The de-modulated signal is fed to head-phone and line amplifiers. Alternatively, the output of the 16 bit codec can be routed to these amplifiers as well.

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LOW PASSFILTER

ANTI-ALIASINGFILTER

16-BITCODEC

OUTPUTCOMPARE

(dsPIC/PIC24H)

DCI & I2C(dsPIC)


SPI(dsPIC/PIC24H) SERIAL FLASH

HEAD-PHONE/LINE

AMPLIFIER HEAD PHONES/LINE

PRE-AMPLIERLINE

MIC

The board also features a serial flash memory device which can be used to store audio data for playback purposes.

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Audio PICTail Plus Board features

Audio Playback using PWM signal from dsPIC/PIC24H Output Compare.

The main feature of the Audio PICTail Plus board is audio playback using the PWM signal generated by the dsPIC/PIC24H Output Compare module. Using this technique, the end application only needs to implement an external low pass filter to able to produce an audio signal. This technique eliminates the need for a high cost codec and is well suited for moderate speech quality applications.

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16 bit Audio Codec

The card features an optional 16 bit Audio codec which allows the user to compare the quality of the PWM playback technique against a 16 bit audio codec. This comparison would allow a user to evaluate the different playback techniques and make the right choice for the end design.

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16 bit Audio Codec 4 Mb Serial Flash

A 4MBit Serial Flash device can be used to store encoded speech samples or other data. The memory interfaces with the dsPIC33F/PIC24H device using the SPI bus.

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16 bit Audio Codec 4 Mb Serial Flash 110mW Headphone Amplifier

A 110mW headphone amplifier drives an external headphone. The amplifier input can be switched between the codec output, the PWM filter output or the input signal. This allows the user to perform a subjective evaluation of the different codec techniques. The amplifier features a digital volume control and the volume is controllable with two push button switches

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16 bit Audio Codec 4 Mb Serial Flash 110mW Headphone Amplifier Microphone and Line input with adjustable gain

The card features a multiple set of inputs and outputs. A microphone and line pre-amplifier is available to boost the input signal to the card. The input signal could be from a condenser microphone or from line outputs of standard audio equipment. The adjustable gain on the amplifier accommodates varying microphone response.

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16 bit Audio Codec 4 Mb Serial Flash 110mW Headphone Amplifier Microphone and Line input with adjustable gain Line output with adjustable gain

Apart from the headphone output, the card features a line output which can be connected to the line input of any audio recording equipment. The line output is driven by an adjustable gain pre-amplifier.

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16 bit Audio Codec 4 Mb Serial Flash 110mW Headphone Amplifier Microphone and Line input with adjustable gain Line output with adjustable gain Works with Explorer 16 and 16-bit 28 pin Starter

Board

The Audio PICTail plus board works primarily with the Explorer 16 development board. It also connects to the 16-bit 28 pin starter board, although certain features of the Audio PICTail board would not be available.

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PWM Technique

We will now discuss the working principle of PWM Audio Playback technique.

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PWM Technique

VIN

T ON

The PWM signal starts of as a constant amplitude square wave whose Ton period is varied in proportion to the audio signal amplitude.

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PWM Technique

VIN

T ON

TON TOFF

PWM Signal

This results in a waveform whose Ton period or pulse width is modulated by the audio signal. On the dsPIC or PIC24H device, this signal is generated by operating the Output Compare module in PWM mode.

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PWM Technique

Duty cycle of carrier varies with input voltage Carrier frequency typically > 4x the sampling frequency 4th Low Pass Filter for demodulating and reconstructing

the audio signal. Lost cost method of audio playback

VIN

T ON

TON TOFF

PWM Signal

LOWPASS FILTER

Audio Signal

The PWM signal is filtered by a low pass filter. This filter is an integrator which produces an output signal which is proportional to the Pulse width of the input signal. The output of the filter is therefore the audio signal which modulated the carrier waveform.

So summarizing the PWM technique,

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PWM Technique

Duty cycle of carrier varies with input voltage

VIN

T ON

TON TOFF

PWM Signal

LOWPASS FILTER

Audio Signal

The Duty cycle of the carrier waveform is varied in proportion to the audio signal amplitude to generate a Pulse Width Modulated signal.

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PWM Technique

Duty cycle of carrier varies with input voltage Carrier frequency typically > 4x the sampling frequency

VIN

T ON

TON TOFF

PWM Signal

LOWPASS FILTER

Audio Signal

The carrier is typically greater than 4 times the audio sampling frequency. So for a 8KHz sampling rate, the application must employ a carrier frequency of at least 32KHz. Using a higher carrier frequency results in better re-construction and simpler filter design.

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PWM Technique


the audio signal.

VIN

T ON

TON TOFF

PWM Signal

LOWPASS FILTER

Audio Signal

A 4th order low pass filter can be used for demodulating and reconstructing the audio signal. The filter can have a pass band frequency of 3300Hz for speech applications.

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PWM Technique


the audio signal. Lost cost method of audio playback

VIN

T ON

TON TOFF

PWM Signal

LOWPASS FILTER

Audio Signal

The PWM Audio playback technique thus presents itself as a low cost solution for moderate quality speech playback applications

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Summary Speech and Audio solutions for a variety of applications.

Summarizing our discussion:

Microchip has Speech and Audio Solutions that fit into many types of applications.

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Summary Speech and Audio solutions for a variety of applications. G.711, G726A can be used for medium to high bit rate

applications.

You could use the G.711 or G.726A coders in applications that emphasize speech quality and accommodate medium to high bit rates

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applications. SPEEX can used for low bit rate applications

You could use SPEEX coder to obtain high compression ratio for low memory and low bit rate speech applications.

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applications. SPEEX can used for low bit rate applications NS, AEC and LEC algorithms for signal processing

The NS, AEC and LEC libraries can be used for front end digital signal processing of speech signals.

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applications. SPEEX can used for low bit rate applications NS, AEC and LEC algorithms for signal processing Speech Recognition for implementing a Speech

command interface

You can leverage the existing user interface of your application by using the Speech Recognition Library for implementing an audio speech command interface.

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applications. SPEEX can used for low bit rate applications NS, AEC and LEC algorithms for signal processing Speech Recognition for implementing a Speech

command interface Audio PICTail Plus Board for testing and implementing

low cost speech playback applications.

And the Audio PICTail Board is a convenient platform to test and develop low cost speech playback based applications.

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References

www.microchip.com dsPIC30F AEC User Guide (DS70134) dsPIC30F NS User Guide (DS70133) dsPIC30F/33F Speech Coding Solutions

User Guide (DS70295)

Here are a list of references for a more detailed understanding of Microchips Speech and Audio solutions offering.

Documents

En 542839Microchip DSC Audio and Speech Solutions