Speech generation
on an Atmel Mega644 in GCC
ECE 4760 Cornell University

Introduction

There are several ways of making a computer talk. The simplest is to record whatever you want to say and play it back at about 8000 samples/sec. The problem with this approach is that it takes a lot of memory and is not very flexible. You could use a dedicated voice recording chip like the ISD chipcorder to record/playback segements of speech. You could also use a speakjet or winbond chip which syntheizes arbitrary speech from fragments of English (called allophones).

It would be cheaper (in hardware cost) to have the MCU directly make speech. One way is the PICtalker approach which is an allophone synthesis software system (based on the obsolete SPO256-AL2 chip). I have ripped the binary allphone file to a matlab program (see below). The main problem with this code is that it requires a 64kbyte table, which would fill all of flash on the Mega644, but could be put into serial dataflash (separate chip). Another problem with this scheme is that the speech designer/programmer has to be very good at stringing together sounds in order to make understandable speech.

Another way is to compress speech so that the MCU can directly do the decompression on the fly. I used differential, pulse-code modulation (DPCM). The motivation for sending samples of the first derivitive of the speech signal, rather than the signal itself, is that the derivitive changes relatively little between samples so fewer bits are required. I implemented a DPCM scheme with 4:1 compression which sends 2-bit derivitive samples. It sounds acceptable, but a little scratchy. I also implemented 8:1 compression (1-bit derivitives). The quality is lower, but still understandable most of the time.

DPCM (2-bit samples)

A version of the DPCM algorithm can be implemented using very little processing time. A 2-bit/sample compressor/decompressor was written in Matlab to encode and to make a packed C header file, and then to do a test-decode. Note that the quantization break-points and reconstruction values are made up by me. You can change them, but you must be consistent in the encoder and decoder. An optimization (program + function) based on the histogram of first derivitives suggests that quantization breakpoints of [-0.05, 0, 0.05] and reconstruction values of [-0.16, -0.026, 0.026, 0.16] are about right for demo wav file given below. A decoder written in GCC for the Mega644 uses the packed code format to generate speech. Each second of speech takes 2 kByte of flash.

To use this system:

1. If you want to have the Mega644 just speak the numerical digits, skip this list and use the code in the next paragraph.
2. Get some clean, noise-free speech. You could record your own voice or use this TextToSpeech demo.
3. Make sure the audio sample rate is 8kHz and save it in a wav file. This little matlab program downsamples a wav file by 2:1. If you use the text-to-speech demo in step (2) you will need to downsample.
4. Run the Matlab compressor on the wav file. The compressor output file will be a table in C header format. You could, of course, have several short compressed tables in flash, or you could index into a long table to say just one word.
5. Resynthesize on Mega644.
   1. Include the compressor output file from step (4) in your c program.
   2. Attach PORTB.3 to a low pass filter, and then to an audio amplifier. The low pass should cutoff at about 18,000 radians/sec (3000 Hz). Sometimes you can skip the lowpass and use the input characteristics of the audio amp to lowpass.

The file DPCMAllDigits.h has a GCC flash array for the digits zero to nine. If you include this in a test program, you have available all the spoken digits. The sample index boundaries for the digits in the array are given below. Using this table you can speak individual digits by decompressing only part of the flash array.

DPCMAllDigits.h is based on the TextToSpeech demo page using the simulated voice "Claire". Commas were placed between the digit names for synthesis. The original synthesis result (wav at 16 Ksamples/sec) and reduced rate result (wav at 8 Ksamples/sec) used as input to the compressor are included for reference.

DPCM (1-bit samples)

A version of the encoder was written that simply sends one bit/sample depending on the sign of the first derivitive. The reconstructed speech has noticably higher noise than the 2-bit version, but is still understandable. The 8Ksample/sec speech waveform (from the TextToSpeech demo page using the simulated voice "Mike") is compressed with a matlab program to produce a C header file, which is included in a mega644 test program. About 60 seconds of speech should fit into flash on a mega644. Attach PORTB.3 to a low pass filter, and then to an audio amplifier. The low pass should cutoff at about 18,000 radians/sec (3000 Hz). Sometimes you can skip the lowpass and use the input characteristics of the audio amp to lowpass.

Allophone synthesis

The code to read and segement out the allophones:

1. The allophone data file
2. the allophone starting point file
3. the program loads the allophone data file and starting point files, then attempts to synthesis "Mega32 speech". Refer to the following allophone description for the meaning of the address segment numbers.
4. SPO256 allophone description. Numbers on the left are used in the matlab code to identify specific allophones.

Using this synthesis style, the challange is to map the predefined 59 allophone sound library into the best approximation of words.

Old Mega32 info is here.

Copyright Cornell University 22-Feb-2010