similar to: Sampling Rate

That's pretty bad. Both DirectSoundCapture and WinMM are capable of recording at 16kHz. I don't know why OpenAL would be incapable of handling it. It's not like it's at all rare or new. I would try 16000 and see if it works. Maybe the docs are wrong? Note that one option to retain high quality is to capture at a higher rate and then downsample using a resampling

What would be speex configuration recomended for Telefone/Voip quality voice? With a quality just a little better/similar then G.729? or GSM? is there a comparison chart somewhere, but telephone quality oriented? Thanks, Alain Tom Grandgent escreveu: > Kirk, > > Speex was designed for 8kHz, 16kHz, and 32kHz sample rates. If you > don't use one of these sample rates,

Oops, CTRL+Enter send strikes again ... At the other end for playback you can convert it back to 48000 (or whatever) by repeating each sample 3 times (48/16 == 3), then running a 8000Hz lowpass over the result to remove any aliasing artifacts. Cheers, David Hogan > -----Original Message----- > From: David Hogan > Sent: Tuesday, 12 December 2006 10:44 AM > To:

Hi, I'm no DSP or audio expert by any means, but I can share what works for me. People in the know, I would appreciate tips on whether this stuff is ok. You could sample at 32000Hz (or 48000Hz, any AC97 card will support this), run a 8000Hz lowpass filter over the data (16000Hz sample rate can only represent frequencies up to 8000Hz) and then drop every second (or 2 out of 3 for

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

Jean-Marc Valin (jean-marc.valin@hermes.usherb.ca) wrote: > > > I've been using Speex in my voice-over-IP program on Win32, in > > wideband (16kHz) mode. I just starting using VBR recently and > > have run into something that might be a problem within Speex: > > Are you turning on DTX in addition to VBR? Also, what version are you > using. As of 1.0, DTX is

Please consider using 16-bit 16kHz (wideband) instead. It's a huge increase in audio quality and the bitrate is still very low, especially if you take advantage of Speex features such as VBR. 8kHz seems totally inappropriate to me for desktop streaming audio, let alone 8-bit samples. Or perhaps your recording equipment is an original Sound Blaster from 1989? (Even that could record at

It seems that I only have the following values available for sampling from the mic. "The value must be 8000, 11025, 22050, 32000, 44100, or 48000" Which leaves 8000 and 32000 for use with speex. I think since this is a game and not a voice application, I'm stuck using the 8kHz rate. What speex setting would you recommend I use for the best quality/performace, what frame size

Hi all, I have recently added Speex support to my app (http://sbooth.org/Max/) and am in the process of creating a GUI for the user to control the codec parameters. I am new to Speex and as such I am not really sure which parameters are more important than others, and deserve prominent placement, etc. Is there a recommended GUI for people to follow or use as an example? Stephen

Jean-Marc, Ok, I understand. It's floating point but I don't have a mechanism in place yet to capture the source data. I'll do that, reproduce the problem, and put together a minimal sample program that can reproduce it given a data file. Thanks, Tom Jean-Marc Valin <Jean-Marc.Valin@USherbrooke.ca> wrote: > > Hi Tom, > > In order to trace the problem,

Hi Reed, I've been using Speex to transmit TTS for years. It works very well with no tweaking. I use Microsoft TTS ("Microsoft Mike") with Speex at 16kHz wideband and VBR quality 6. Sometimes I forget that the sound is even coming from another computer and being compressed+decompressed. If anything, TTS seems easier for Speex to deal with than real voice. But I don't

I have (finally) posted my Speex wrapper classes. They are at: http://www.grandgent.com/spx/tgAudioCodec.zip I followed your recommendations and they worked fine with 1.1.0. However, I'm still having the same problem with 1.1.7 that I had the last time I tried to upgrade. I'm using the same code with both versions, except for calling speex_encode_int instead of speex_encode, and

Josh, Yes, you can change VBR quality or ABR average bitrate at any time. You don't need to do anything special. I use this feature (changing VBR quality midstream) from time to time and it's quite convenient. Tom Joshua Gargus <schwa at fastmail.us> wrote: > > Hi there (from speex-dev newbie), > > Is it possible to change the target VBR on the fly? The

Ok, I figured it out. :) This seems to work: 1) Call speex_bits_read_from() once, specifying the location in memory of the compressed data, and the total length of that data. 2) Keep calling speex_decode() until speex_bits_remaining() returns 0. Then you don't have to keep track of the # of frames per packet, or the size of each compressed frame. It's done magically by the codec.

Hi, I've been using Speex in my voice-over-IP program on Win32, in wideband (16kHz) mode. I just starting using VBR recently and have run into something that might be a problem within Speex: If someone hasn't spoken for a little while, and the bitrate drops to very low, sometimes the high half of the spectrum becomes frozen with a looping sound. The bottom half of the spectrum is

----- 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:

Hi all, Please find below an updated Speex Internet Draft document. It would be good if we could book some time for discussion on Speex at the IETF meeting in Vienna (scheduled for 14th July). The cutoff for submission is 9:00am EDT, (GMT -04:00), 30th June. Comments and feedback welcomed! Regards Phil

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

hello, i have a simple question about speex frame size. i work with jspeex - but i think it's speex generic, the question i have. can i use any frame size i want? or does the size have be in a certain ratio to other numbers? and what does the frame size in the decoder mean? encoded or decoded packet frame size? how much data can i hand to the encoder to encode? if i hand i.e. 400 bytes to

Frame size of 320 means 320 samples, which is 640 bytes of data if the samples are shorts. Speex can work with samples as shorts or floats - for example, see speex_encode_int vs. speex_encode. In both cases the values should be signed ranging from -32768 to 32767. I suggest trying the sampleenc.c and sampledec.c programs in the doc directory. If those work, then maybe you can spot what