similar to: sampling rate

Hello, We currently have code for sampling at 8 KHz at device layer. In order to have minimal code change while still doing wideband I am thinking of packing two 20 ms frames of the low band data as input to Wideband mode for encoding. Is this a feasible idea? I actually tried it out. Speech came out at the decoding end, but I could occasionally hear low pitched, high frequency

Hi, I was wondering if anyone has experimented with Speex's wideband (16kHz) mode at lower quality settings. In particular I have been using quality 3, and with wideband input files the resultant frequency spectrum is limited to about an upper end around 3.5kHz (almost telephony quality bandwidth). Has anyone tried increasing the spectral bandwidth at the expense of lowering the

It seems to work ok with the same audible quality as a standard sampling rate. Is there any way to test this? Will superimposing an inverse wave over the origional produce a meaningfull result? Thanks for your time, Ryan de Leeuw <p><p>>Sorry for the delay. I've been doing a couple tests >and what I'd suggest >is encoding using the narrowband (8 kHz normally)

Hi! Having had a look at speex I have understood that the primary aim is to provide speech compression for low-bandwidth things such as VoIP and whatever. Also, the fact that the highest frequency (wb mode) is 16kHz makes me wonder if... The REAL question: Would speex be useful for storing the voices for characters in a game or for storing other voice recordings with high quality? I do not know

Hi, I just wanted to know if there are any recommendations for pre-/post-processing (processing power isn't a question). I'm aiming at 16kHz and tried an lowpass-filter at 11kHz before encoding but this didn't improve the results... <p>Regards, Thomas --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe

> The cost of down-sampling, if done efficiently, is probably less then > the cost difference between 32 kHz and 44.1 kHz so it's probably worth > it. If you don't care about standard sampling rate, you could even to a > 2/3 conversion which would get you 29.4 kHz... I'm curious why not just sample at a lower rate if it's just VoIP anyway? My opinion is that 44kHz

Hello there I've developed a p2p voice application using Speex and I'm looking for ways to reduce Speex's cpu usage. My K6-2 300 MHz can't even encode 16 bit audio at 16KHz in realtime using Speex in narrowband mode. I've tried to lower the quality to 2 and complexity to 2 also but it's still way too slow. Which other ways are there to make encoding faster? Is there a

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

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

hello, I have recently compiled speexenc for WinCE/StrongARM SA-1110-processor. It seems to run with no errors, however the encoding time is very slow - 10 seconds of speech takes aprox 3 minutes to encode. Even though the target architecture is significantly less powerfull than a standard PC, I am trying to figure out why it is this slow. Does speex use floatpoint calculations? The input raw

Hi, We would like to announce the first release of the Speex project. Speex (http://speex.sourceforge.net) is an open-source (LGPL), patent-free compression format allowing an alternative to expensive proprietary codecs. Unlike Ogg Vorbis which compresses general audio, Speex is designed especially for speech. For that reason, Speex is meant to be a complement to Vorbis. Since it is specialized

> Date: Fri, 05 Dec 2003 13:22:53 -0500 > From: Jean-Marc Valin <Jean-Marc.Valin@USherbrooke.ca> > > I think I see what you mean, though I haven't been able to listen to > your wma file (not everyone has a wma decoder). The problem probably > only lies in the VBR tuning for wideband which hasn't received much work > yet. One way to check that is to encode in

Hi all, I've got a 400MHz XScale-PXA255 board, and I want to stream voice from it over a network connection at 28.8baud. This calls for a capable voice encoder which can encode at about 24kbps. I was damn happy when I found Speex and said goodbye to MP3 :) However, i'm still a long way from realtime encoding using speexenc, is this possible? Using the fixed point math option in

At present my stream is at 11.025 kHz and I want it to be at 44.1 kHz. Input is coming from line-in on my sound blaster card under linux (RH 9) using the sb driver. I presume that it is icecast that sets the sample rate on the dsp in the card, though when I change the settings in icecast.xml and darkice.cfg as show below the stream becomes choppy; or rather the sampling doesn't seem to

2 things, first an idea... next a question. QUALITY MEASUREMENT IDEA: I find it difficult to hear 2 voice samples and tell which is nearer the original, especially if the background hiss is slightly different. So what if you actually subtract the post-compression sound from the original and then listen to the DIFFERENCE. If you can't hear any voice except background noise and some hiss from

OK. So I am running a 1.5 Ghz P4 with 256mb runing redhat 7.3 using Liveice, Lame and Icecast. Liveice and lame on the source box and icecast on the streamer box. 96k and 56k streams sound wonderful. But the 24k stream sounds like gerbils talking to each other. Any ideas of what would be causing this? Thank You, Mike <p> -- ///////////////////////////////////////// - Mike

Hi all, I'm trying to use the FLAC C libraries to encode audio. I'm doing something like: FLAC__seekable_stream_encoder_set_channels(pflac->fse, 1); FLAC__seekable_stream_encoder_set_sample_rate(pflac->fse, 11025); FLAC__seekable_stream_encoder_set_bits_per_sample(pflac->fse, 8); if ((bps = FLAC__seekable_stream_encoder_init(pflac->fse)) !=

Hello Jean-Marc >If you want to do it, I can show you >what functions (there are 2-3) to port. Otherwise I might do it >eventually, but it's not a top priority (there's already an SSE version >though). I would indeed like to know which functions can be used to improve K6-2 performance through 3DNow. Cheers Bjoern D. Rasmussen <p><p><p>>From: Jean-Marc

Is there a list somewhere of "standard" encoding rates? I know, for example, CDs are encoded at 44100, as is a lot of digital sound, but I've seen programs that specify different levels of quality (like radio, phone, tape, CD) and I'd like to know if there are some encoding rates that are accepted as standardized for recording at different levels of quality. If so, is there

FYI, the Asterisk software PBX <http://www.asterisk.org/> has now incorporated my recent patches to support dynamic RTP payload types. As a consequence, its SIP implementation now supports Speex, so if you have a Speex-compatible SIP client, you can use it to make calls using Asterisk. Some caveats: - Only narrowband (8 kHz) Speex is currently supported; not wideband. (Unfortunately,