similar to: added background noise problem?

Using narrow, wideband, and ultra-wideband encoding on a short 16khz wav gave .spx's of 3,789 ... 2,935 ... and 1,875 bytes. Even after reading the manual, smaller files for the higher frequency encoding seems counter-intuitive. My mp3 at 32 kbps on the original 22khz wav is 3,866 with a quality comparable to speex wideband on the converted 16khz wav, so speex is a 24% improvement in size.

Hi, Sorry to be posting about a subject that may have already been answered. If so, please point me in the right direction. I'm developing a dictation application on the Dell Axim (Windows Mobile 5.0 Pocket PC). A key requirement of the application is the best possible sampling rate as the audio goes into a speech reco system. So, I've set up my wrapper around libspeex to capture audio

First of all, thank you for your input Tim. That is very helpful. I would love to hear from other people with experience of AEC and Speex. I guess I have to split my question into to parts now. 1. Is it a fact that using the windows multimedia API (wave audio) for audio capture and playback makes it impossible to do echo cancellation with Speex AEC or other EC method due to inprecise timing? I

I know the focus for Opus is low delay, but I've been watching its development with interest because of the potential for audiobook/podcast use, where latency is practically irrelevant. I hear the upcoming USAC codec will give good results for this niche (though listening test results don't seem to be available to the public yet), but I also hear it'll be extremely patent

Dear List, I've been looking tirelessly for a solution to this dilemma but without success. Perhaps someone has an idea that will guide me in the right direction. Suppose I have the following data.frame: DF = data.frame(X = c(114.5508, 114.6468, 114.6596, 114.6957, 114.6828, 114.8903, 114.9519, 114.8842, 114.8579, 114.8489), Y = c(47.14094, 46.98874, 46.91235, 46.88265, 46.80584, 46.67022,

hi on my device i can sample only at 32khz and want to use speex at 16khz so i need to down-convert the input signal by factor of 2. does anyone provide me a reference to some code that does that? are there any trick to do that? i tried to add to subsequent sample but the result was very bad. what are the requrment from a decimation filter for audio? thanks, nir -------------- next part

----- Original Message ----- From: "Alexander Chemeris" <Alexander.Chemeris at sipez.com> To: "Vincent Burel" <vincent.burel at vb-audio.com> Cc: "Conrad Parker" <conrad at metadecks.org>; <speex-dev at xiph.org>; "Jean-Marc Valin" <jean-marc.valin at usherbrooke.ca> Sent: Thursday, November 13, 2008 11:31 PM Subject: Re:

Hello Mailing List, I am a Speex supporter and user that would really like to know how much memory Speex uses to decode a 8kHz, 16kHz and 32kHz (primarily the 8kHz) and is it possible to use a 1kBytes of RAM to decode a 8kHz stream? (I was thinking of the possibility of using a ATmega168 to decode Speex) //P?r, Sweden

Kirk, Speex was designed for 8kHz, 16kHz, and 32kHz sample rates. If you don't use one of these sample rates, you'll be messing up important assumptions deep within the codec. Why these sample rates? It's telecommunications tradition, rather than PC audio tradition. If you want an efficient and high quality format for voice chat, try 16kHz with VBR quality 6. You should see

On Fri, 6 Apr 2001, John Griffiths wrote: > ok so 24kbps for 56k modems... > > can i go any lower and get the 28 k modems? (still a lot of them about) or will 24 be good enough fo that? As others have said, 16kbps should do the trick. Keep in mind though that the quality of the sound will also depend on the sampling rate. MP3 will handle some higher sampling rates higher than some of

The background noise is unacceptable. I'll have to stick with 32kbps mp3s eventhough the files are 70% larger.

This is a really good point, and definitely a recurring theme on this mailing list. :) I wonder, what are some better options for handling this issue, other than to keep saying "just use 8/16/32kHz"? - Extend Speex to support other sample rates (seems unlikely..?) - Integrate a resampling algorithm into libspeex - Maintain a list of recommended resampling libraries that work well

('binary' encoding is not supported, stored as-is) Hello! First let me say that after several experiments i have done today, encoding a wav to several ogg bitrates and then listening to them in winamp, i'm realy pleased with the sound quality of the final release 1.0 of Vorbis. But during these tests i've done, i found that OggDrop XPd crashes when i try to encode at 16KHz, 22KHz

Hi list, I am researcher in VoIP Applications and my challenge now is convert one RTP data frame that is in 32KHz to other RTP data frame in 32KHz. Do someone help me about it? Very thanks, Thiago. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/speex-dev/attachments/20090316/fbbaf566/attachment.htm

I found the details: 44100hz, 16bits, stereo I am looking at rewriting the program to record at 8Khz/16Khz/32Khz mono only should I record at 44100hz (and convert down) or record at the required hz level? ----- Original Message ----- From: <speex-dev-request@xiph.org> To: <speex-dev@xiph.org> Sent: Thursday, June 14, 2007 3:44 AM Subject: Speex-dev Digest, Vol 37, Issue 19 >

Hi! I'm using the newest (beta2) speex library on PocketPC (Windows powered) and experiencing some problems with noise suppression preprocessor turned on while encoding 44kHz files. You can hear the example here: http://szalik.net/speex (this is a 44kHz, 16, mono file encoded in UWB mode) I tried using speex cross-compiled with gcc (fixed point + arm4 asm) and VS (just fixed point) and it

Hi all ! If someone could give me a hint on how to proceed with the following i'd be very happy: I have a test setup on an nrf52832 (Cortex M4) in which I receive audio from a PDM microphone (64 sample frame) and pass it directly to an I2S device i.e. from ISR to ISR. With uncompressed audio this works just fine. Now I try to insert OPUS1.3 in the path but cannot make it work. The

Hi there, i'm not into speex internals but i feel that your problems are due to the fact that speex doesn't do yet fixed point operations and it seems that you lack FPU on your PDA. So i would expect a big boost in performances when the fixed point version of Speex will be out. Cheers, Luca On 8/20/06, Sunil <dualmax88@yahoo.com> wrote: > > Hi, > > Sorry to be

All -- Does anyone have experience doing remote live broadcasts over Icecast? My thought is to use a Dell laptop running Windows (yeah, I know ;-), digitize locally to 16khz, and pump the output to a remote Linux box. Has anyone done something like this before? Thoughts? Issues? Thanks, Roy --- >8 ---- List archives: http://www.xiph.org/archives/ icecast project homepage:

Hi, In my dial plan I've configured two trunks to make outbound calls (trunk1 and trunk2) to same service provider but I want when any of my exten starts with _2. should goto trunk2 and there should be some kind of disturbance (like some noise or some background noise) when my calls goes to trunk2 to make the call quality bad. Mainly I want to achieve bad call quality on trunk2 by adding