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Comment: Re:Whats the point of ultra low bitrate codecs? (Score 1) 179

by drowe67 (#33657930) Attached to: Codec2 — an Open Source, Low-Bandwidth Voice Codec
Correct for VOIP, the main target for codec2 is digital radio where the same overheads don't apply. For VOIP there are some tricks - if you concatenate many channels in a single IP packet (say for trunking between sites) you could send 32 calls in the same bandwidth as a single 64 kbit/s channel. For Voice over 2.4GHz Wifi you could consider breaking 802.11. For example the minimum bit rate is currently 1 Mbit/s. That could carry 500 x 2000bps calls if the protocol was modified. As it's unlicensed spectrum this is possible and legal, like running cordless phones and toys on the same spectrum. Alternatively we could come up with a 2000 bps Wifi waveform and get a 26dB power advantage for longer range, non line of site etc.

Comment: Re:English only ? (Score 1) 179

by drowe67 (#33657840) Attached to: Codec2 — an Open Source, Low-Bandwidth Voice Codec
The tones in Chinese are short term variations in pitch. It's not really that different to the way we use pitch to convey emotion and questions in English, although perhaps the variations are faster. Codec2 explicitly analyses and encodes pitch. So it should be fine. I'm learning Mandarin myself so will do some tests with "Wo de LaoShi" (my teacher) soon :-)

Comment: Re:Packet loss? (Score 1) 179

by drowe67 (#33657794) Attached to: Codec2 — an Open Source, Low-Bandwidth Voice Codec
if we include an "erasure mode" this type of codec is pretty good at handling packet loss, as it is easy to interpolate between two adjacent frames. CELP type codecs have a lot more memory so tend to be less robust. Also conversational speech has only about a 30% activity factor, so 7/10 packet losses will be in silence of background noise frames.

Comment: Re:what about LATENCY? (Score 1) 179

by drowe67 (#33657666) Attached to: Codec2 — an Open Source, Low-Bandwidth Voice Codec
David Rowe, the author here. The latency is about 40ms. The encoder accepts buffers of 20ms (160 samples) and the decoder outputs buffers of 20ms (160 samples) So assuming zero transmission delay, you get your first output speech sample about 40ms. It's comparable to cell phone codecs like GSM, and fine for real time communications.

Comment: Re:How does it handle background noise? (Score 1) 179

by drowe67 (#33657480) Attached to: Codec2 — an Open Source, Low-Bandwidth Voice Codec
This is an area of codec2 I would like to work on. Can you please send me some sample files of fireman speech corrupted by background saw noise? This would be a good start. The good news is with an open source codec this problem can be addressed - with a closed source codec your are stuck.
Communications

+ - Freest Phone Calls Ever?

Submitted by
drowe67
drowe67 writes "Li Yuqian (Beijing, China) and I (Adelaide, South Australia) just made the first VOIP call using the IP04 Open Hardware IP-PBX. Unlike any other PBX projects the IP04 hardware is free (as in speech). Anyone is welcome to copy, modify, manufacture and hack the design. The hardware was designed using open source gEDA CAD software, and it even runs uClinux and Asterisk. Could these be the freest phone calls ever? Even the inventor of the telephone Alexander Graham Bell was wrapped up in 19th century patent wars over his hardware!"

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