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Old 30.01.2002, 00:25   #1
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Exclamation Work experience with sound.

Hi pipl,

I'm the 1st time here. Glad to be in your forum.

Here is the problem:<BR>Has anyone here worked with sound (played wav, ..., etc.), in all aspects, especially in field of sound compression algorithms and sending sound in net?

Thx.
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Old 30.01.2002, 04:03   #2
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Welcome aboard Talex.

Actually my job is related also to that sort of things <IMG SRC="smile.gif" border="0">

The moderator of this section professionally knows sound compression stuff.

Transfering sound accross the net is also not too far from us, so go ahead and ask questions if you've got any.
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Old 30.01.2002, 19:13   #3
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2acid: thx for reply.

2 acid and moderator

i've a lot's of questions <IMG SRC="smile.gif" border="0">

here is some of them:

1. i'm reading .wav file all is ok, except - double buffering. i think u already understand now why i have asked about buffers. How could i organize this? As u said me about buffers or this is another case? - reading data from file in one buffer and playing data from another buffer, in separate threads. i'll try do do as u said.

2. what special things except sending by tcp or udp (i don't know which to use) packets/datagrams for example of wav file to the remote host, and play it there. If wav compressed using PCM. Maybe some other thing i've to do. By the way u know about ITU-T organization and their recommendations - G711(PCM), G721, G723,...? Now i'm trying to send wave file to remote user and play it at him.

thx
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Old 31.01.2002, 01:35   #4
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OK let's start.<BR>1. Yea as I mentioned before, threads gonna work for that.

<BR>2. Normally, ITU voice compression protocols have to be used with RTP protocol, which is using UDP as transport layer protocol. That's how most of the IP phones are transfering their voice data once connection has been established with another kind of protocol like SIP or H323. The original voice data stream is dividing into frames, it is small parts consisting of 160, 240 or other amount of samples, then each frame will be compressing according to protocol specifications and then will be placed into UDP packet.

But of course you can design you software to transfer you voice data across the network using TCP protocol. In my opinion it would not be appropriate if the network quality is low.

Try to read more of IP telephony.
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Old 05.02.2002, 23:26   #5
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Please, any links to SIP, H323 and RTP protocols!

thx
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Old 06.02.2002, 01:12   #6
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http://www.ietf.org/, кликни на RFC pages, там техническая документация.

http://www.sipforum.org/
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