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Yes but I need to encode at 2Kbps.. are there other free codecs expecially projected for very-low but rate? ---------- Original Message ---------------------------------- From: Jean-Marc Valin <Jean-Marc.Valin@USherbrooke.ca> Date: Tue, 30 May 2006 00:51:25 +1000 >Le lundi 29 mai 2006 ? 16:45 +0200, Leonardi Andrea a ?crit : >> Hi! >> Is it possible to encode at a rate of

Hi to everyone. I'm using speex in an application at the minimum bit rate (2.15 kbps), setting the encoder quality at 0, but obviously the quality is terrible. My question is: there are some signal processing that I can do to improve the quality of the signal at very low bit rate? Or can you tell me other codec that works well at that rate? Thank you.

Thanks for the reply. We are currently using AMBE (4Kbps) for our Traffic/Weather Channels. If you have ever had a chance to hear the service, you will know that AMBE does not do us well. I understand that 2Kbps is low quality, but any poorer than AMBE? If can get a decent quality for other low bandwidth talk channels, such as about 10-16Kbps and have it sound rather clean, then I would be

I'm not sure, but I think that Microsoft have the license now.. anyone know if there is the source code to download? Or others codecs.. thank you very much ---------- Original Message ---------------------------------- From: "John Miles" <jmiles@pop.net> Date: Mon, 19 Jun 2006 10:15:59 -0700 >Do a search on "Voxware MetaVoice". I'm not sure who licenses

Hi, <p>apparently, I didn't delete my test samples ... I found them. :) Here's a small table of what nominal bitrate fits to which quality at what sample rate. Quality levels inbetween -1 and 0 are possible, but I didn't test them. Nominal bitrates of those quality settings will be somewhere between those of -1 and 0 and therefore can be guessed. q/channels || -1/mono |

Thanks for the answer. So data are stored in little-endian format on the WAVE file. But have I to encode in big endian format or little endian is ok?! Thanks ---------- Original Message ---------------------------------- From: "John Miles" <jmiles@pop.net> Date: Tue, 23 May 2006 12:15:05 -0700 >You are probably reading little-endian samples on a big-endian platform, and

On Sun, 29 Aug 2004 17:53:29 -0700, Ralph Giles wrote: > On Mon, Aug 30, 2004 at 03:03:28AM +0100, Andy Baxter wrote: > >> Is there any way to bring the bitrate in ices2 down below 32 kbps? > > Generally the trick for this is to downsample the audio before encoding. > You can ask ices to do this with a resample stanza in the config file: > > <resample>

Thank you. You're right ... my error ... I meant to say 12 bytes (including the 2 bytes for VAD). And it is 10ms/frame. No matter ... thank you for the SPEEX specs. In terms of quality, what SPEEX bit rate compares with G.729 at 8kbps data rate please? Is there some reason why you chose the 20ms frame rate? Do you keep that same frame rate for the different bit rates? The faster frame

Thank you Jean-Marc. My understanding is that G.729 is a telephone codec, so there must have been some reason why its developers went to 10ms/frame. Do you know why that might be? From a recent post on this list I saw somebody talking about your decoded sample rate being 8KHZ/sec. and then he mentioned that being 160 bytes at 20ms/frame. That said, I take it that your decoded samples are

I'm having a issue with build libThusnelda 20081201,video is blocky, you can clearly see video is blocky in some parts of the video. the Bit rate is high so what is the problem is, on x264 this type of Bit rate will make the video very smooth. can anybody tell me how to get that smooth video and what settings can i use, I'm interested in the advanced usage of Thusnelda with two pass

Is there any way to bring the bitrate in ices2 down below 32 kbps? I'm setting up a demo for someone of how to use linux to do net radio broadcasting. The setup I'm thinking of is to use ardour plus jack to mix two (maybe more) input sources (live audio and recorded tracks/programmes), then send the mixed audio to ices2 for streaming to icecast2, using the jackified version of ices2. This

Is there any way to bring the bitrate in ices2 down below 32 kbps? I'm setting up a demo for someone of how to use linux to do net radio broadcasting. The setup I'm thinking of is to use ardour plus jack to mix two (maybe more) input sources (live audio and recorded tracks/programmes), then send the mixed audio to ices2 for streaming to icecast2, using the jackified version of ices2. This

Hi, > You can try the Ogg DirectShow filter to get Windows support for Speex: > http://www.illiminable.com/ogg/ if you wait a little longer, I am trying to provide speex support for the ffmpeg project (www.ffmpeg.org). It does encode/decode to/from avi and wav, a support for ogg has not been done, and my question is, should it be necessary? I don't really like ogg in first place..

I'm sorry, but I've found in the paper "Speex: A Free Codec For Free Speech" at page 5 the phrase "If more than one channel is used, then an instance is required for each channel (Speex is not stateless)". What does it mean? Could it be that is the problem that I've met? Thanks Andrea ---------- Original Message ---------------------------------- From:

Hi, I am looking how to setup speex codec in codecs.conf for 8 Kbps and 6 Kbps. In config file are many parameters for setting. I don't know what is need to change for narrowbad like 8 Kbps and 6 kbps. Any suggestion? [speex] ;0-10 quality => 4 ;0-10 complexity => 4 ; true / false enhancement => true ; true / false vad => false ; true / false vbr => false ; 0 = off, otherwise,

Hallo, yesterday I was puzzled when I discovered that I probabliy miss something in the interepretation of intercept in two-way lm models. I thought that the intercept, using the default contr.treatment contrasts, represents the mean of the group of observations having zero in all column of the model.matrix. It turns out not to be case To be more more clear I am attaching a short example:

Hi, I am new to speex and intend to port Speex to an SH3 based Single board Application computer running at ~100 MHz. The application scenario is for intercom kind of an Application. I want to know if the CPU power on the SH3 based SBC running at ~ 100 MHz shall be sufficient for running speex(The compression and decompression interfaces shall be used simultaneously by the application).

This mail depends upon the fact that I don't have a couple of good earphones ;-) I read in the site that q=6 is a very high quality, but does it contain perceivable differencies from the original? (for 95% of people, of course). I also found q=6 to produce files slightly bigger (1/10 bigger) than those produced with lame VBR q=2 (about 192 bps on average). I always thought LAME VBR q=2

Hi all! I installed icecast & ices (from CVS) Enerything runs, but with some problems: When I connect to the server (with XMMS or MPlayer) the buffer runs empty very quickly, causing interruptions. My "server" is an old Pentium 120 with 64MB RAM ,a HD of 8G and an ethernet NIC. (I know it isn't much, but I'm tired of taking my CD collection to work.) I ripped the tracks in

Hello, thank you for responding. I'm sure I'm confusing something, but I did not explicitly set either kHz or kbps, I just set the mode to narrowband, and traced through libspeex's code to see that sample_rate was 8000 something, I'm guessing Hz. My assumption was: if you set PA to nb-mode, then you should encode 160 bytes to 21. Is that mistaken? ys On 6/13/07, Jean-Marc Valin