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Open Source Codec Encodes Voice Into Only 700 Bits Per Second (rowetel.com)
Longtime Slashdot reader Bruce Perens writes: David Rowe VK5DGR has been working on ultra-low-bandwidth digital voice codecs for years, and his latest quest has been to come up with a digital codec that would compete well with single-sideband modulation used by ham contesters to score the longest-distance communications using HF radio. A new codec records clear, but not hi-fi, voice in 700 bits per second -- that's 88 bytes per second. Connected to an already-existing Open Source digital modem, it might beat SSB. Obviously there are other uses for recording voice at ultra-low-bandwidth. Many smartphones could record your voice for your entire life using their existing storage. A single IP packet could carry 15 seconds of speech. Ultra-low-bandwidth codecs don't help conventional VoIP, though. The payload size for low-latency voice is only a few bytes, and the packet overhead will be at least 10 times that size.
I wonder how it performs on tonal languages like Cantonese.
http://michaelsmith.id.au
I've been way thing for a new cold deck for joyce recordings.
That's starting to approach feeding the sentence into a speech to text system at one end and then sending the text over the air to be fed back into a text to speed converter.
I read the internet for the articles.
The samples don't sound great, and I really wonder how well it does trying to record a conversation rather than one person talking directly into a mic. Still, I would welcome the chance to try an app based on this to see if it could really record your day, although until I can test it I'm a disbeliever.
I'm an American. I love this country and the freedoms that we used to have.
A new codec records clear, but not hi-fi, voice in 700 bits per second -- that's 88 bytes per second.
It's 87.5 bytes/s and it's that odd 1/2 byte that keeps it from being too fuzzy sounding for hi-fi.
It must have been something you assimilated. . . .
Encoding voice more efficiently has implications far exceeding the amount of storage space required to save it. There's a reason why the article is comparing the new codec to single sideband. When transmitting digital data over radio, it pretty much invariably (nowadays) means some sort of spread spectrum transmission. The fewer bits required per second means the less spectrum you are having to spread your signal over, this the more concentrated your signal is. A radio transmitter has a fixed power output, so if you are smearing that power over less band, then you have a stronger signal.
It is a testament to the amateur radio pioneers of the past that an analog radio transmission mode invented over a hundred years ago is, just now, being possibly rivaled in its efficiency.
They're skirting the bottom edge of comprehensibility, the voice in the samples is by no means "clear". You have to focus very closely to understand that is being said much of the time, and even then, repeated listenings are sometimes necessary.
To compute the channel capacity, you need to know the channel's signal-to-noise ratio as well as its bandwidth.
The Shannon channel capacity formula is: C = B * log_2(1 + SNR) where C is the channel's capacity in bits/second, B is its bandwidth in hertz, log_2 is the base-2 logarithm and SNR is the channel's signal-to-noise ratio.
If we assume an SNR of 48 dB for a reasonable POTS line, its capacity would be C = 3 kHz * log_2(1 + 48 dB) ~= 3000 * log_2(63097) which is almost 48,000 bits per second.
This is a theoretical limit that realizable systems can only approach, but never equal or exceed. A practical system would also use extra bits for forward error correction purposes; I doubt that this codec deals gracefully with bit errors.
For back-of-the-envelope purposes, assume you could use this codec to send a single voice signal in 700 Hz of bandwidth on a channel with low SNR, or you could send 60 voice signals over a regular POTS line.
Actually, our modems degrade gracefully. The least-protected bits go wrong with low bit-error rates, and the more protected bits survive. It takes a high bit error rate to kill it. So bit errors result in the speech being "off" but not dropping out.
Bruce Perens.
I've been programming all day, and haven't said many words at all. There are people who talk for their entire work day, but they generally spend half their time listening and more processing something, so they may actually do 4 hours of speech or less in the work day. Most people don't really speak for more than a few hours per day.
Bruce Perens.
Here's a link to the current source code, as it wasn't straight forward to find: https://svn.code.sf.net/p/free...
Licensed under GNU LGPL v2.1.
This space is not for rent.
That's who'll be interested in technology like this. They could compress and store the conversations of every person in the U.S., 24/7/365, for decades, without having to upgrade their data storage capacity.
Just to show I'm not all gloom-and-doom: I'd think NASA, and private spaceflight companies like SpaceX, would be interested, since a low datarate for voice communications would be great, I'd think, for interplanetary distances. With higher datarates available you could have multiple conversations happening simultaneously.
It's free software, not for sale.
Bruce Perens.
Those samples are anything but "clear." It's still impressive, given the compression ratio, but there's no need to go overboard. You wouldn't want to have to rely on your understanding of one of these samples
systemd is Roko's Basilisk.
Or trump could yell at someone in a tweet
Nullius in verba
I guess it's impressive to get anything other than straight noise out of less than 1kbps. But I've wondered why Rowe hasn't focused more on quality at more moderate (e.g. 2-3kbps) bitrates rather than continuing to seek ways to trade away some quality for an ever lower bitrate. It's been a couple years since I tried it out and came to that conclusion; this looks like that trend has continued.
I couldn't get my encoded samples to sound nearly as good as the samples posted on the codec2 site. And it seemed like the second-lowest bitrate at the time (1400?) sounded essentially just as good as the highest (3200), which meant it wasn't making effective use of the additional bits. The quality jump between its highest mode and the lowest Opus mode (at 6kbps) was huge . (EVS would be a big jump over that.)
From what I understand, codec2's most prominent competition operates at 2.4kbps and up and sounds noticeably better at those rates than codec2 does.
It's not for recording.
It's for giving us Voice communication to MARS and back. If you have the ability to transmit voice over long distances using lower bandwidth, you can add in luxuries like checksums and redundant data so that when you send it a very long distance it arrive at the extreme distance away where your 10,000 watt transmission is weaker than a dollar store walkie talkie.
Ham radio is where most of the breakthroughs in communication happen. I can see this mode used to allow voice communication with mars astronauts. We already have PSK31 allowing a ham with 2.5 watts of power to transmit text messages around the globe easily.
Do not look at laser with remaining good eye.
I am very glad that fight is over. And as far as I can tell, we saved Amateur Radio entirely. It would have died in our lifetimes.
Bruce Perens.