Displaying 13 results from an estimated 13 matches for "codec2".
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2011 Jul 12
1
Speex newbie: win32 encoder and Java applet playback?
...me
command line encoder (if possible for win32) to re-compress the files,
and then use a Java based solution (an applet would be just fine) to
reproduce these low-bitrate audio files from a web page.
2. I have found several low-bitrate codecs, namely g.729, and another
open source one dubbed "Codec2". I wasn?t able to find a command line,
win32 encoder for these... much less a java player applet solution
that I can embed on web pages.
a. Codec2, at http://www.voiptoday.org/index.php?option=com_content&view=article&id=512:codec2-low-bit-rate-open-source-speech-codec-v01-alpha-rele...
2016 Aug 26
2
Using opus on ATMEL 32-bit RISC microcontroller
...nk the stream enoguht?)
>
> This is going to be a problem. Assuming you mean 8 mega*byte* (and not 8
> megabit), that's still only 2 kilobit/second. Opus pretty much requires
> 8 kb/s, so 4 times what you have. The only codec I know that can do 2
> kb/s with reasonable quality is codec2, and the implementation is
> floating point.
>
> > Since I'm using FIXED_POINT, I have to pass also --disable-float-api?
>
> Yes. The normal API has calls with both int and float, so if you compile
> with FIXED_POINT, the float calls do a conversion to int before using
>...
2014 Nov 04
2
Opus vs Speex NB
Hi,
I noticed that speex.org has a banner that mentions that Opus is better
than Speex in all aspects. The supported bitrate range for Speex seems to
be as low as 2kbps though but Opus can only go as low as 6kbps. Is this one
aspect where Speex is still preferred? (I understand that it's not a very
common scenario though).
Thanks,
Manpreet.
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2016 Aug 26
0
Using opus on ATMEL 32-bit RISC microcontroller
...Sorry for my typo, the flash size is 8MB (Byte) :-)
Unfortunately I cannot use another flash because I'm working on a
proprietary board.
Jean-Marc, thanks for your suggestions.
I thought to use fixed point for convenience, but I can work on floating
point too, so I will take in account the codec2 (I didn't know it).
Moreover, if you all have other suggestions to give me I appreciate! ;-)
Cheers,
Daniele.
Il 2016-08-26 18:12 Amit Ashara ha scritto:
> Hello Daniele
>
> It would be worthwhile to attach an external serial flash or USB thumb
> drive, if the intent is store d...
2016 Aug 26
3
Using opus on ATMEL 32-bit RISC microcontroller
Hi Jean-Marc,
thanks a lot for your reply.
> Well, the first question is whether you want encoding, decoding, or
> both. If there's one you don't need then you can remove that
> (unfortunately, there's no easy way right now).
You're right! I forgot to say that I need only the encoder side (and
only for voice).
My application have to acquire a 16bit 8KHz PCM stream and
2015 Mar 18
2
Asterisk 13. Writing call quality parameters to CDR. How?
Hello.
Voice quality when calling - this is one of the most important in the PBX.
You need to record the quality parameters for each call to improve.
Because the overall quality of a call can only be determined upon
completion, I did it in the HangUp handler and wrote in custom fields of
CDR.
This worked well in asterisk 11.
In asterisk 13 I did not find a handler after the call, but before
2015 Mar 19
0
Asterisk 13. Writing call quality parameters to CDR. How?
...rom_u` varchar(30) CHARACTER SET utf8 COLLATE utf8_czech_ci NOT NULL,
`uri` varchar(30) CHARACTER SET utf8 COLLATE utf8_czech_ci NOT NULL,
`useragent` varchar(30) CHARACTER SET utf8 COLLATE utf8_czech_ci NOT
NULL,
`codec1` varchar(10) CHARACTER SET utf8 COLLATE utf8_czech_ci NOT NULL,
`codec2` varchar(10) CHARACTER SET utf8 COLLATE utf8_czech_ci NOT NULL,
`llp` varchar(10) CHARACTER SET utf8 COLLATE utf8_czech_ci NOT NULL
COMMENT 'lost packets by local end',
`rlp` varchar(10) CHARACTER SET utf8 COLLATE utf8_czech_ci NOT NULL
COMMENT 'lost packets by remote end ',...
2014 Nov 05
0
Opus vs Speex NB
For anything below 6 kb/s, I would actually recommend using codec2
rather than Speex or Opus.
Cheers,
Jean-Marc
On 04/11/14 05:55 PM, Manpreet Singh wrote:
> Hi,
>
> I noticed that speex.org <http://speex.org> has a banner that mentions
> that Opus is better than Speex in all aspects. The supported bitrate
> range for Speex seems to be as...
2016 Aug 26
0
Using opus on ATMEL 32-bit RISC microcontroller
..., do you think opus shrink the stream enoguht?)
This is going to be a problem. Assuming you mean 8 mega*byte* (and not 8
megabit), that's still only 2 kilobit/second. Opus pretty much requires
8 kb/s, so 4 times what you have. The only codec I know that can do 2
kb/s with reasonable quality is codec2, and the implementation is
floating point.
> Since I'm using FIXED_POINT, I have to pass also --disable-float-api?
Yes. The normal API has calls with both int and float, so if you compile
with FIXED_POINT, the float calls do a conversion to int before using
int internally. If you disable t...
2015 Sep 18
1
Speex and low bandwidth communication
Hi,
I'm fairly new on codecs, I'm trying to implement a communication
between two PCs.
The data rate should be of approx. 6Kbits/s, this 6Kbits/s can just be
the useful data. Any encapsulation (for example UDP , RTP, etc ...) can
be present even if that means to rise the overall bit rate.
Also if this rate can be achieved on Asterisk. For exemple I've tried to
change de rate on
2004 Jan 19
3
configuration to Grandstream via tftp
Hi,
Anyone know how to set up tftp server for grandstream.
I gues it should be somethink like
<tftpserver-dir>
<mac-address>
firmware.bin
config.txt
Is this correct ?
And how should the config-file look like. ?
I had search sipphone.com but did'nt find anything.
/HHA
_________________________________________________________________
Rethink your
2013 May 22
1
is it possible to bring speed below 1000 bit/s
Hi folks,
I am totally new to audio streaming codecs, and just looking around. Trying to
figure out what else I can put on top of my very long list of projects.
So a few days ago I figured out that fldigi a digimode application for Amateur
Radio supports a dual 1000 Baud PSK mode. I thought "that's fast" and started
looking around which sort of Data I could put into that.
A
2011 Oct 11
11
Reporting for Asterisk Call Center
Dear Tariq;
About elastix.org, this can be use with Asterisk or it is coming as a complete IP Telephony, Call Center, IVR and Reporting?
Because, I do not need to install another IP Telephony on the server which already has asterisk which is an IP Telephony, this will cause a problem in the service (for example, when listening for SIP port of 5060).