similar to: Real time audio encoding - cpu usage

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

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

I was also wondering if there is a standard set of input sequences people are using to test Speex. I haven't stumbled upon it/them yet. > -----Original Message----- > From: owner-speex-dev@xiph.org [mailto:owner-speex-dev@xiph.org]On > Behalf Of Jean-Marc Valin > Sent: Tuesday, February 25, 2003 7:24 PM > To: speex > Subject: Re: [speex-dev] Quality > > > > I

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 Greg If money isn't a problem Intel has an optimized compiler for eVC and XScale processors http://www.intel.com/software/products/compilers/techtopics/PCA_Optimization_WP.pdf If you have any luck getting the eVC compiler closer to realtime I'd really like to know. I'm still far from realtime when using Speex 1.1.3 on a HP iPAQ (Intel pxa255). Best regards Bjoern D.

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 complexity) and some wideband modes also work (up to ~20 kbps) at

Hi, I'm not sure if I should post this question to this list. If not; please tell me. Ok, here it comes: Is the following code correct for compressing audio? The output I get is so extremely small, but what is more important: if I pass it through zlib, it gets at least 50% smaller! Init: int bitrate = max_bandwidth * 8; speex_bits_init(&libspeex_bits); if

Le mar 18/11/2003 à 17:39, Allen Drennan a écrit : > Speaking of video conferencing in conjunction with Speex, we are > currently beta testing a solution we developed that offers multi-point, > multi-party video and audio using the Speex engine for voice. > > http://www.wiredred.com/downloads/ecsetup.exe > > The fair and good audio settings are Speex narrowband, high quality

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,

Dear all, i am currently evaluating which codec to be used for a "very narrowband" voice communication system (using either VoIP either GSM CSD) on mobile devices running Windows Mobile and Symbian OS. I had experience in using AMR-NB 4.75 with VAD on Symbian OS and all worked very well. However, AMR it's not royalty free and if i need to use it on Windows Mobile i will need

> Ok, I forgot to mention that I want to incorporate speex into my own > program (which is for e.g. streaming audio and a couple of other > things). JM> One thing you'll need to decide: do you want to pack in into an Ogg JM> container or will you take care of handling the packets yourself? I take care of that myself. JM> If you JM> want to pack it in Ogg, I suggest you

Hi, I am working with libspeex api. I would like compare the size of speex file encoded with libspeex and the size of speex file encoded with speexenc. For example, I encode with speexenc (narrowband, quality 1) a wav file of 1.43 MB (17 seconds) output a spx file weight 21KB, for other hand I encode with libspeex the same wav file with same characteristics in arm processor, pocket pc and the

> What is the maximum frame-length that libspeex will produce for narrow, > wide and ultrawide encoding? In normal operation (no in-band side information, like requests, ack, stereo, ...), the max size for a frame is 62 bytes in narrowband, 106 bytes for wideband and 110 bytes for ultra-wideband. Jean-Marc -- Jean-Marc Valin, M.Sc.A. LABORIUS (http://www.gel.usherb.ca/laborius)

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

It looks like I'll need to go further into the guts of speex to do this. I do, however, see some lines in nb_celp.c/nb_decode() that look interesting. I guess I'll play with it. I doubt that it will be terribly clean, though. Jean-Marc: Take a look at line 1195 in nb_celp.c (CVS). It reads "speex_warning ("More than to wideband layers found: corrupted

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

At 07:55 PM 3/11/03, Jean-Marc Valin wrote: > > - Only narrowband (8 kHz) Speex is currently supported; not > > wideband. (Unfortunately, the assumption that audio sample rate == 8 kHz > > is riddled throughout the Asterisk code.) > >Perhaps it's still possible to send wideband, while telling Asterisk >it's narrowband (the bit-stream is such that you can decode

Benny Prijono a ?crit : > So my settings now (I use embeded Visual C++ 4 latest SP): > - plain C (not ARM assembly), fixed point version of speex (r12832) > - quality/complexity set to 5 (previously it was 0-1). 3-4 is also > okay, but obviously quality is not as good. > - compiler set to optimize for maximum speed (/Oxt flag, previously > it was default, which is mapped to

Le jeu 15/01/2004 à 15:30, Daniel Vogel a écrit : > Unrelated, but please use SSE/MMX/... intrinsics on Windows instead of using > inline assembly so you also get the speed benefit on Win64. OK, so here's a first start. I've translated to intrinsics the asm I sent 1-2 days ago. The result is about 5% slower than the pure asm approach, so it's not too bad (SSE asm is 2x faster

Hello I'm working about audio and video codecs. I'm making a research about the state-of-the-art in coding technologies. My question is about the speex codec 1.0. I would like to know the name of the algorithms that it uses.(LPC,CELP..). It uses more or less the same algorithm for narrowband and wideband? Finally i would like to ask about future codec developement projects. Thanks