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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.
Can this be used for two-way comms? conversion time from analog to the bitstream, across the net and converted back to voice, what's the delay?
Slashdot's name? When my compiler sees
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.
70 years * 365 days (roughly) * 24 * 60 * 60 * 88 bytes/sec / 1024 / 1024 / 1024 = 181GB
Is my math off or are they assuming such people will only have a 15 year life span?
VTT is a trivially solved problem, is it? Especially at low latency on embedded devices?
The words you're looking for are "I'm sorry"
15s/IP packet - this should lower operational cost for our government.
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.
Good old POTS had 3k of audio bandwidth. What is the bandwidth of this CODEC? It's hard to be impressed without knowing the details.
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.
May be boring, but it's great for espionage!
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.
It m___ cer___ly c_n!
T__s is just th_ thing Telco_ and oth_r _____ prov___rs need to _ed__e usag_ and all__ more users __ lim_ted bandw__th circ__ts.
He__. C_n y__ call m_ bac_ on my house__one?
" A single IP packet could carry 15 seconds of speec"
great
A stream of sounds is difficult to parse. Converting it via various codecs won't change that or make it more useful. Converting the analog wave sounds into meaningful digital data (in the form of words as text, musical notation, specific fart parameters, a database of whale or bird calls, etc) is more helpful and efficient. Meaning can be extracted and/or analyzed. As someone else suggests, those can be converted back to a semblance of the original sequential stream of sounds (but why?).
If you are communicating with a person who has a particularly melodious voice, you may want to preserve the analog, but not the 88Bps version.
...omphaloskepsis often...
Do we finally have a 2400b mode? Would love to do digital but when existing FM transceivers. Due to HOA I can't (and yes have tried) do HF reliably.
. Define sqrt(x) as something really evil like (x / rand()), and bury it deep. Watch your coworkers go nuts.
> [I] haven't said many words at all.
And this is from a guy who is famous largely for saying stuff!* Well known for talking about Morse code, talking about free software and open source, talking about Debian's principles, talking at conferences, probably talking to Congress ... and even you don't talk more than a few hours per week.
* and also of course for DOING a lot of things, including doing things like founding organizations - which requires a lot of talking.
Actually, that got me curious, what do you first / most really got your name out there, why do you start getting so much press attention? Busybox is important, of course, but you never hear the person who created grub mentioned in press, or the original author of glibc.
A couple of typos made that hard to read. Let me try again:
What do you think first / most really got your name out there?
Why did you start getting so much press attention, etc, compared to other people who also did important work?
Not that you aren't worth listening to. I'm not saying you don't "deserve" the attention or whatever. I'd just like to know your thoughts on how and why someone like yourself becomes a bit of a celebrity in the field.
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.
Also a bit optimistic about battery life.
More likely the NSA can use this to store everything you say forever.
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.
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.
Thanks for that. Sounds like I have a lot of work to do to become nerd famous. ;)
I just checked out your blog and found the bit about switching power supplies interesting. I knew about switching *regulators*, but didn't realize common power supplies could actually run on DC. I'll have to check your blog more often.
Or trump could yell at someone in a tweet
Nullius in verba
I wonder if Google could pair Codec 2 700c and RAISR (Rapid and Accurate Super Image Resolution) for YouTube videos that use even lower bandwidth than the 144p that exist already. Or, they could use the same technology to reduce the bandwidth necessary to stream 1080p/4k/8k videos and further embarrass the data capping ISPs.
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.
The jump in intelligibility and voice quality going from 4kHz narrowband to 6kHz mediumband is big- probably bigger than going from mediumband to 20kHz fullband. The distinguishing features of many consonants are between 3.5 and 6 kHz.
Finding some way to take advantage of information beyond narrowband - even if not trying to encode much of it - could be a distinct advantage for a low bitrate codec over existing competition.
They're not that bad. It's a human voice codec, and for more than half the samples I could tell that I was probably listening to a human voice. All you'd need to add is subtitles so you can tell what's being said and it'd be pretty good.
And now for a less facetious reply: I've seen something like this before when working on audio processing algorithms that are evaluated subjectively, after hearing the same type of audio sample for the millionth time in a row, and having heard 900,000 less-good versions, you start to think that it's sounding pretty decent. It isn't until you either measure it objectively or find a fresh test subject who's never listened to any of the previous attempts to listen to it and provide a subjective rating that you realise it's actually not so good. There's a technical name for this problem which escapes me at the moment... anyone?
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.
Homer Dudley had a working vocoder pre-WW2, which was used in the encrypted voice system SIGSALY.
From Wiki, this encoded voice into 12 signals, each with 6 levels (call it 2.5 bits) at 25 Hz. That's about 750 bits/s.
if you need subtitles then it is by definition not fucking pretty good.
What a weird summary:
The new codec isn't "competing with single-sideband modulation".
Normal SSB is unprocessed speech. So the codec is simply competing with natural speech.
The claim that SSB "is used by ham contesters to score the longest-distance communications using HF radio" is just plain wacky. So they use natural speech too talk to each other???
Is it a form of adaption? I couldn't understand the 700C samples on the first few play throughs, but after 5 repetitions they made sense.
Slashdot: where don knuth is an idiot because he cant grasp the awesome power of php
This call for an implementation on those ESP8266 and similar modules: ADC and DAC (or PWM if absent) to interface with headset and that codec to send voice over IP sparing most possible bandwidth for other data and/or degraded link conditions.
Also an Arduino or other cheap platform and a couple serial rf modules could be an interesting way to tinker with the protocol and explore applications.
Will this run on a 6502 or more importantly is this what bender uses?