similar to: Sampling Rate

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,

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

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

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:

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

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

hello, Is there any future or current work being planed on other sampling rates (besides 8kHz and 16kHz), specifically 11026Hz ? <p>Ryan <p><p><p>__________________________________________________ Do you Yahoo!? Faith Hill - Exclusive Performances, Videos & More http://faith.yahoo.com --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage:

Hello everyone I've a following problem with the speex codec. I'm developing application that is supposed to read sounds from the microphone with DirectSoundCapture. Unfortunately when I want to compress and decompress the sound buffer from DirectSound I only have jitter. Maybe you have some code samples how to compress standard DS buffers: LPDIRECTSOUND8 lpDS = NULL;

Hi, is there any way to tell the decoder the output sampling Fz we want ? opus_decoder_create = Sampling rate of input signal (Hz) Considering this example (VoIP-out from WebRTC/RTP) MICROPHONE(44.1/48kHz) >> [encoder created at 48kHz but with internalSampleRate set to 8kHz]>> INTERNET >> [decoder(created with 48kHz)] >> 48kHz(?) >> G.711(8kHz) This leaves us with

The encoder and decoder can handle, 8, 12, 16, 24 and 48 kHz input/output. If doesn't matter what it gets encoded to/decoded from. you can initialize a decoder at 8 kHz and it'll still decode 48 kHz audio fine (you just won't get the high frequencies obviously). For sampling rates other than 8/12/16/24/48, then you'll have to do resampling. Have a look at the speexdsp resampler if

Hi Julien, Quote from : http://dspguru.com/dsp/faqs/multirate/resampling "The problem is that for resampling factors close to 1.0, the interpolation factor can be quite large. For example, in the case described above of changing from the sampling rate from 48 kHz to 44.1 kHz, the ratio is only 0.91875, yet the interpolation factor is 147!" My guess is that Opus would perform similar to

Kemal, It sounds like you are doing something wrong. I strongly recommend that you profile your application to see exactly where the CPU time is being spent. AMD happens to have a nice profiler called CodeAnalyst that they give away for free. And it's plenty usable on Intel CPUs as well. http://www.amd.com/us-en/Processors/DevelopWithAMD/0,,30_2252_3604,00.html Make sure you test a

On Thu, 2007-03-29 at 12:53 -0700, Brian Willoughby wrote: > > Hello FLAC list. > > > > As far as I know 24 bit FLAC support is broken. It often doesn't > > compress the audio at all, but instead stores the chunks as verbatim > > type (although the FLAC format supports 24 bit). Perhaps this is > fixed? > > If so, do let me know. > > Hi

Hello, is somebody working on a good low-sampling rate / low-bitrate mode? I encoded today a mono/16KHz/16bit WAV (a TV-talkshow), using OggDrop. The quality of the '64kbps' mode was unacceptable, so I had to use '80kbps' mode. The bitrate averages around 42 kbps, which I found a bit high for this quality. In your opinion, what bitrate should I expect as Vorbis matures? 24 kbps?

Hi, I am trying to use speex for online speech compression in my application code. I wanted to know the bits per sample for 8khz sampling rate. Presently i am using 16 bits/ sample, is it alright? -- Bhushan

Hi, another question on the same topic Speex resampler at 44.1kHz seems to be very CPU intensive on Android (even more than the Opus encoder) While Speex at 48kHz is just fine. I wonder any alternate solutions or ideas ? Improve it, look for alternate solution ... I am guessing the NEON optimization are still used for both, etc. On Thu, Apr 2, 2015 at 4:46 PM, Jean-Marc Valin <jmvalin at

Hi, i get the next problem when i try with Alsa sound Driver. I get a sound on a lot of parts... OS: GNU/Linux Ubuntu 6.10 Linux maximi89-desktop 2.6.17-50-386 #2 Tue Jan 23 16:48:16 UTC 2007 i686 GNU/Linux 00:04.0 Multimedia audio controller: nVidia Corporation CK804 AC'97 Audio Controller (rev a2) maximi89@maximi89-desktop:~$ WINEDEBUG=err wine '/home/maximi89/.wine/drive_c/Archivos

Jean-Marc Valin wrote: > Hi, > > I just released unstable version 1.1.2 that contains more fixed-point > work. Though it's still not 100% complete, enough have been done to make > it run in real-time on ARM. In order to do that, compile with > --enable-fixed-point --enable-arm-asm. All narrowband modes work in > real-time with complexity 1 (some work with higher

Hi, I have a link_to_remote and it update a div. After the first click the div is opened with the information. Now I want to be able to close hide the div . How should I do it. Someone at the IRC told me to use toggle. But what I want is a bit odd. I want that it will not do again link_to_remote but toggle. Tnx Kfir

Hi All. I am seeing the following unexpected behavior with OPUS_SET_MAX_BANDWIDTH. I expect that setting this to OPUS_BANDWIDTH_NARROWBAND would give similar results to passing an 8Khz sample rate stream, but OPUS_SET_MAX_BANDWIDTH has almost no effect with any settings. My test data has 4Khz bandwidth. I am testing the opus encoder (latest versions) with the following opus_encoder_ctl