search for: 8kbs

...secting a professional's code. I work for a company that is currently using a G729A vocoder from a 3rd party software company and is looking into speex so they no longer have to pay royalties. The product we are trying to force speex into is based on a TI C5416 DSP that did narrowband 8-bit, 8kbs. The product was fairly full as it is so some modifications had to be made in order to fit speex into the project just to allow it to link. The modifications are based off assumptions that I made when looking over the code so I may have been absolutely wrong. The main assumption was about the e...

...d hear it and I could understand it! unfortunately the quality is so poor it sounds like a cylon, and I know speex is much better than this because I have a windows version that works and the quality is quite high (by quality I mean how it sounds, both systems are using a quality of 4 with 8KHz and 8kbs voice). I think I've narrowed down where the problem is coming from but I have no idea why it is happening. my problem is that when I go to decode my data the decoder doesn't do anything a lot of the time because it goes here: if (speex_bits_remaining(bits)<5) re...

...d hear it and I could understand it! unfortunately the quality is so poor it sounds like a cylon, and I know speex is much better than this because I have a windows version that works and the quality is quite high (by quality I mean how it sounds, both systems are using a quality of 4 with 8KHz and 8kbs voice). I think I've narrowed down where the problem is coming from but I have no idea why it is happening. my problem is that when I go to decode my data the decoder doesn't do anything a lot of the time because it goes here: if (speex_bits_remaining(bits)<5)...

...d hear it and I could understand it! unfortunately the quality is so poor it sounds like a cylon, and I know speex is much better than this because I have a windows version that works and the quality is quite high (by quality I mean how it sounds, both systems are using a quality of 4 with 8KHz and 8kbs voice). I think I've narrowed down where the problem is coming from but I have no idea why it is happening. my problem is that when I go to decode my data the decoder doesn't do anything a lot of the time because it goes here: if (speex_bits_remaining(bits)<5) re...

...name is Vasily. I am a software developer from Ukrane, and i am going to use Speex on TMS320F28335 DSP-controller wich has only 68KB of RAM. And I worry about this. I can't find memory requirements in speex manual(only some about optimization). Can anyone tell me, how many RAM Speex can eat at 8kbs realtime encoding and decoding with echo canceller and jitter buffer enabled? P.S. I found some about echo canceller, but nothing about the encoder.

Depends what you want to do! Do you want to do VoIP over that T1 to a provider or IP telephones? Do you want to hook up to the PSTN through that T1 as 24 voice channels, through a T1 card on your asterisk? If you want to use the T1 as 24 voice channels, the Telco is going to have to re-provision the T1 as a voice T1, because currently, presumably it is one big channel of data. You could have

I swear I read somewhere at sometime a command line that someone put forth that used mpg123 and sox to normalize the MP3 for MoH to a constant bitrate, etc... Through my search again for this information I can't find it. Can someone tell me if I am crazy and if so... does anyone have an "off the top of their head" solution to getting the bitrate constant for MoH? Thanks, Mike

hi together, we using icecast 2.2.0 with suse linux 9.1 and mp3pro. i have many problems with icecast, because im finding no way to changing dj?s without lost a part auf our listners. the option <client-timeout>45</client-timeout> dosent work. now at dj changing, icecast kick all listners, when the source diconnected. we have test the fallback option, here the winamp users dont move

Hello, I currently have speex "working" in my project but the encode and the decode don't seem to be working like I would hope. I am using speex 1.2beta2 and I am trying to 16bit 8kbs narrowband. on the decode side I encoded a voice on a windows machine. I took the encoded information and transferred it over to the C54x project and decoded it and it didn't sound that great. I watched the decode process and it would appear that the first 40 of the 160 words (2 bytes) are...

...hard to > be more specific. A potential gotcha I just discovered: iTunes is slightly confusing as regards its MP3 encoding bitrates: if you encode in mono, you need to specify the 'Stereo Encoding Bitrate" as TWICE the mono bitrate you want! Our mono theme music has been running at 8kbs rather than the 16kbps I thought it was, which is why OUR fallback stream has been dropping every time our live feed connects.

...---------------- So here's the relevant details of our plan. The following is subject to change - and any constructive suggestions for change will be seriously considered. 1) We're going to write a Codec using the DSP starter kit that will convert speech at 8/16 kbps 16 bits to/from an 8kbs serial stream. 2) We will some Linux utilities which will allow the TI developer kit to be connected to ethernet etc. This connection would require the use of one of the standard daughter cards for TI DSP kits - or you could build your own (I'm going to) using the adapter plugs and a couple...

...here's the relevant details of our plan. The following is subject to > change - and any constructive suggestions for change will be seriously > considered. > > 1) We're going to write a Codec using the DSP starter kit that will > convert speech at 8/16 kbps 16 bits to/from an 8kbs serial stream. > > 2) We will some Linux utilities which will allow the TI developer kit to > be connected to ethernet etc. This connection would require the use of > one of the standard daughter cards for TI DSP kits - or you could build > your own (I'm going to) using the adapt...