similar to: How suitable is speex for high-quality speech?

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

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

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

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:

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

Hello all, I'm testing the Speex codec for my diploma thesis on a BF-533 Blackfin under uCLinux (2005R3 RC3 release). I successfully compiled the Speex (1.1.11-svn) and I can encode/decode wav-files on my STAMP-board using the speexenc/speexdec sample apps. But I encountered that the decoded file sounds strange/noisy, when compiling with "--enable-blackfin-asm" +

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

> 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

I brought this up 12 months ago or more. I believe the low-pass filter defaults too high with quality -1 in Vorbis 1.0. Is this going to change in the future? I think it should at least cut off at 15khz not 16khz and perhaps even 14khz. I believe it's important when streaming. Removing some of the higher frequencies compared to removing more of the audible sound makes more sense to me.

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

> 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

>> you are better off using the vogg orbis codec. speex is meant >> specifically for telephonic voice. it takes a single human voice and >> compresses it well. it cannot handle muliple voices or music very well. >That part is true, so of course it depends on the application. I guess I >should have added that for most applications, 16 kHz is recommended >instead of 44.1

I'm trying to get speex to encode a bit faster, mainly by rewriting a few functions in SSE and translating the GCC __asm__ to VC __asm. There's 2 functions I'm targeting, first is vq_nbest which consumes 40% of the time at high complexity and split_cb_search_shape_sign. Which consumes just over 30%. I've split out two functions from: cb_search_precompute_energy - loop at the

Hi, recently I made some test with the opus tools (enc and dec) and I'm very (and positively) surprised about the resultant quality. But the only think that I miss is the ability to change the low pass filter frequency via "--lowpass" option or similar. For example at a quality or 96 kbps the cut off of the filter starts at 16Khz and is completely cut at 20 Khz. But in case of

hi are there any codecs around that allows high quality as in "studio lite"? it may consume high bandwidth, and hopefully allow some packet loss. roy

Hi Jean-Marc, I thought at quality 3 (wideband) - wb_submode1 that the 4-8k band was not using a codebook table. From the code I can see that some sort of "lsp" encoding is performed. What exactly does this encode? (I assume lsp means line-spectral pairs) The reason I am asking is I'm comparing the "effective" spectral bandwidth of Speex against the

Hello, congratulations to the Ogg Vorbis team - RC2 sounds good. But... RC2 in 128 kbps mode seems to clip off all frequencys beyond 16 KHz. On the tracks I tested Beta 4 gave response even beyond 18 KHz. Some testings on a randomly chosen track: (other tracks gave similar results) Artist: Judas Priest Album: Jugulator Title: Bullet Train Beta4: 127 kbps, ~ 18 KHz (!) RC2: 132 kbps (!), ~ 16

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

Hello all. I've been playing around with theora since it first entered CVS, and I like what I'm seeing. Today I've been fixing xine's theoraplugin to understand theora_info.frame_{width,height} and theora_info.offset_{x,y}. I only got it working after some experimenting and basically copying the code from player_example. A few questions related to this: 1. What are the legal

Hi, I have ported the speex codec to PocketPC. However there are some issues coming. Firstly, i am trying to use a sampling rate of 44.1KHz and 14KHz after changing some of the code. When i try to decode and listen , there is some break in sound which occurs only at the starting of the sample file. On the first run, it works fine but the break increases upto a certain level on subsequent runs .