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Codec2 — an Open Source, Low-Bandwidth Voice Codec
Bruce Perens writes "Codec2 is an Open Source digital voice codec for low-bandwidth applications, in its first Alpha release. Currently it can encode 3.75 seconds of clear speech in 1050 bytes, and there are opportunities to code in additional compression that will further reduce its bandwidth. The main developer is David Rowe, who also worked on Speex. Originally designed for Amateur Radio, both via sound-card software modems on HF radio and as an alternative to the proprietary voice codec presently used in D-STAR, the codec is probably also useful for telephony at a fraction of current bandwidths. The algorithm is based on papers from the 1980s, and is intended to be unencumbered by valid unexpired patent claims. The license is LGPL2. The project is seeking developers for testing in applications, algorithmic improvement, conversion to fixed-point, and coding to be more suitable
for embedded systems."
I'll be presenting on Codec2 at the ARRL/TAPR Digital Communications Conference this weekend in Vancouver Washington, Near Portland. I'll try to get the video online.
Bruce Perens.
The original rationale for Codec2 is at Codec2.org. I've been promoting this issue for about four years, as I was bothered by the proprietary nature of the AMBE codec in D-STAR. But I didn't have the math, etc., to do the work myself. It was really fortunate that David became motivated to do the work without charge. He has a Ph.D. in voice coding. By the way, look over his web site rowetel.com for the other work he's done: two really nice Open Hardware projects - a PBX and a mesh telephony device, an Open Source echo canceler for digital telephony, used in Asterisk and elsewhere, and his own electric car conversion. He'd be my nomination for the MacArthur grant.
Bruce Perens.
It is a real-time codec on my workstation and is intended to be a real-time codec on embedded DSP. It's currently all floating point and does things it should not like malloc of multiple buffers per sample
Download the code and build it. It's "just type make" on Linux. The raw (uncompressed) sample format we've used for testing is 16-bit samples at 8 KHz and there are some tools to play those, and some pre-recorded samples. Not too much trouble to figure out.
Bruce Perens.
Right. Sorry. Real time on the x86 workstation I'm using. Not converted to fixed-point for weaker CPUs yet. Not tested on ARM, Blackfin, AVR, etc. Waiting for you to do that :-) Downloadable code. Reasonably portable. Type make and let fly.
Bruce Perens.
Jean-Marc Valin is on the project mailing list and David is another Speex developer and the person Jean-Marc recommended to me. We are trying for an improvement over Speex at low rates.
Bruce Perens.
Speex isn't great in this application, because at low bitrates there is a significant delay through the codec and the output stream requires far too much bandwidth to be useful. Consider that digital speech systems like Mototrbo, TETRA, P25 and Iridium typically have less than 6kbps throughput once you've taken FEC into account.
Congratulations on the license, OM. We haven't yet explored how to wedge this into D-STAR, but sending it as data rather than voice would be one way. All of the D-STAR radios except the latest one, the IC-92AD, use a plug-in daughter board to hold the AMBE chip, and it might be that somebody could make a dual-chip version of this board sometime. Since AMBE is proprietary we are stuck using their chip if we want to be compatible, unless the repeater does the conversion for us using a DV-Dongle. They sell TI DSP chips with their program burned in, and don't give out the algorithm.
It may be that on D-STAR the AMBE chip also does the modulation for a data transmission, just doesn't run the codec. But the modulation is known and there is a sound-card software implementation of D-STAR that interoperates with it. I don't have any D-STAR equipment to test. The folks on dstar_development@yahoogroups.com know a lot more about D-STAR.
73
K6BP
Bruce Perens.
The repeater can rebroadcast the data, but that data would be AMBE encoded, and AMBE is both trade-secret in its implementation and patented in some of its algorithms. There may be an AMBE chip in the repeater, I've not played with one. The usual way one converts to and from AMBE on a PC is with a device called the DV-Dongle, which contains the AMBE chip. This costs lots of money and is not nearly so powerful as the CPU of the computer it's plugged into, which is one reason to be fed up with proprietary codecs.
So, if you had some newer, Codec2-based radios, and some older D-STAR radios, linking repeaters might be a good way to get them to talk to each other.
This is hand-waving about a lot of issues, like we've not designed the next generation of data radio to put Codec2 into. One might guess that such a thing could use IPV6, and better modulation than just FM, and FEC, etc.
Bruce Perens.
you'll be happy to know that it's a fellow Australian ham developing this Codec2 - David Rowe, VK5DGR Here's a link to David's development page
It encoded those 3.75 seconds in 0.06 seconds and decoded in 0.04 seconds on my AMD Phenom 9750 2.4 GHz, one core only, compiled with GCC and the -O3 switch. That's all of the overhead of the program starting and exiting, too. It's using floating, not fixed point.
This, it seems, bodes well for low latency of the final implementation on a DSP chip.
Bruce Perens.
We don't know yet, but I don't see how it could be worse than AMBE in D-STAR, which makes various eructions when faced with large packet loss. I did various sorts of bit-error injection inadvertently while debugging yesterday, and right now you still get comprehensible voice with significant corruption of the LSP data. This, IMO, indicates an opportunity for more compression. Handling the problems of the radio link is more a problem for forward error correction, etc.
Bruce Perens.
There are currently 51 bits in a frame. That is the minimum that you can send, and you'd send 40 of those per second as the codec is presently implemented. A real data radio would add bandwidth for its data encapsulation, but would have to meet the time and bandwidth requirements of the codec payload.
Bruce Perens.
The DSP Innovations codec manages decent speech quality at 600bps, god knows how (proprietary closed source). I think this the state-of-the-art in low bitrate codecs just now.
"I bless every day that I continue to live, for every day is pure profit."
I have been working on mod_codec2.c for FreeSWITCH, which is committed in a WIP module. The library for codec2 isn't a library at all just yet. I'm working with David and Bruce to make sure we can get a working libcodec2 in place ASAP so we have a real VoIP demo that people can compile, call and test against. /b