Voice compression

CVSD uses a bit-serial data stream. Harris datasheets for obsolete Codecs are HC55516, HC55532. The "recording"-circuit can be an analog hack ( Kop, flipflop, 4 Bit shiftregister ) that sends data via SPI. The "playback" would have to emulate this circuit in software and output via a 8 bit D/A ( R2R resistor network, but serial ICs may be easier in SMD ).

MfG JRD

Il 02/04/2025 19:55, Rafael Deliano ha scritto:

Thank you for the suggestion. I tried to implement a simple CVSD codec in Python just to test the quality. I finally got these two functions[1].

I started from this audio[2] and obtained this one[3] after an encoding and decoding process. It's a short speech from an italian voice. I think you can see how bad the quality of decoded audio is.

I suspect I made some errors, because I don't think this is the quality of this audio codec. You said this codec was used in the past, but even if the quality some years ago wasn't high, the quality I reached in my implementation is very poor, quite unusable.

My sounds is quite clear, they are generated by a TTS engine. Then they are flashed on the chip memory.

encoded_stream = bytearray() encoded_byte = "" last_bits = 0x00 for sample in samples: bit = 1 if sample >= prev_sample else 0

# Aggiorna il valore del campione precedente if bit == 1: prev_sample += step_size else: prev_sample -= step_size

# Adatta la dimensione dello step guardando gli ultimi 3 bit last_bits = last_bits << 1 last_bits += 1 if bit == 1 else 0 last_bits &= 0x07 if last_bits == 0x00 or last_bits == 0x07: step_size = step_size * 2 else: step_size = step_size // 2 # Limita la dimensione del passo if step_size > STEP_SIZE_MAX: step_size = STEP_SIZE_MAX elif step_size < STEP_SIZE_MIN: step_size = STEP_SIZE_MIN

encoded_byte += "1" if bit == 1 else "0" if len(encoded_byte) == 8: encoded_stream += bytes([int(encoded_byte,2)]) encoded_byte = ""

return encoded_stream

def cvsd_decode(bitstream): prev_sample = 0 step_size = 16 STEP_SIZE_MIN = 16 STEP_SIZE_MAX = 16384

samples = [] last_bits = 0x00 for byte in bitstream: for sbit in f"{byte:08b}": bit = 1 if sbit == "1" else 0 if bit == 1: prev_sample += step_size else: prev_sample -= step_size

samples += [prev_sample]

# Adatta la dimensione dello step guardando gli ultimi 3 bit last_bits = last_bits << 1 last_bits += 1 if bit == 1 else 0 last_bits &= 0x07 if last_bits == 0x00 or last_bits == 0x07: step_size = step_size * 2 else: step_size = step_size // 2 # Limita la dimensione del passo if step_size > STEP_SIZE_MAX: step_size = STEP_SIZE_MAX elif step_size < STEP_SIZE_MIN: step_size = STEP_SIZE_MIN

return samples

Try 6.5 khz. I'll write a little more later but I've dealt with this problem and there are some reasonable approaches.

You are not seriously expecting me to debug your code ?

CVSD 16kBit was used in the 70ies for military secure communication. The then SpaceShuttle ADM ( = CVSD ) is a simple digital implementation,

The CML FX209 is an early integrated analog version:

The Harris HC55564 is a simple digital IC.

The CML FX609 is the next and final generation with PCM-like filter that reduces high frequency noise.

We did use the Harris. On switching to the FX609 had a test with all the employees in the company with handset what they liked better: 90:10 for the FX609. The problem with "better" is that everyone is accustomed to PCM-filtered speech.

All these ICs one can get via ebay.com from China. Using 2 on breadboards one can build a simple channel that "distorts" speech for testing. Reference would be an old PCM-chip.

As for quality: the 64kBit PCM may be the gold standard, but the cordless DECT phones use ADPCM at 32kBit with hardly any loss of quality. This is not the original CCITT-ADPCM that was very complex. But i still doubt implementation on an AVR is easy.

CVSDs i did years/decades ago on PICs/68HC05.

MfG JRD

Yes, reducing a little the sampling freq is a good solution.

From 8kHz to 6kHz the quality stays acceptable and the bitrate decreases from 32kbps to 24kbps.

Il 05/04/2025 11:12, Rafael Deliano ha scritto:

I didn't write this.

From your last post, it wasn't clear to me if CVSD was a real suggested solution for my application. I tried but the quality is very bad (if my code is ok). Isn't a solution for me for the audio quality we are used to these days.

It's much better to reduce the sampling frequency from 8kHz to 6kHz without touching anything else in the ADPCM playback system.