similar to: Speex test cases?

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

On Tue, Feb 18, 2003 at 06:09:43PM -0500, Jean-Marc Valin wrote: > Le mar 18/02/2003 ? 17:38, John Hayes a ?crit : > > If I encode something in ultra-wideband, can I decode it in wideband by > > chopping off bytes in every frame? > > All you have to do is use the --force-wb switch with speexdec. It will > decode as if the file were wideband, ignoring the ultra-wideband

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

I have a patch for libspeex, which optimises some of the loops in vq_nbest and vq_nbest_sign that speeds up encoding - my results: test file: 10s wav file at 16000 Hz, mono encoding with wideband --quality 3, --comp 3 machine: PIII-900Mhz, 256MB RAM before: 2.78s after: 2.38s I'm still trying to grasp the code (I'm just a coder, no background in sound processing), and just optimised

If I encode something in ultra-wideband, can I decode it in wideband by chopping off bytes in every frame? John --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list, send a message to 'speex-dev-request@xiph.org' containing only the word 'unsubscribe' in the body. No subject is needed.

Hello, Using asterisk-1.2.1 I am trying to convert my music-on-hold files from .wav to alaw: % sox moh.wav -r 8000 -c 1 moh.al resample -ql The file sounds fine when listened with: % sox mox.al -t ossdsp /dev/dsp But when listened through asterisk with an alaw SIP phone the sound is clicky and too fast. Did I forget something in my conversion command? -- ldm@apartia.fr

On Tue, Feb 18, 2003 at 09:06:16PM -0500, Jean-Marc Valin wrote: > BTW, when you have something working and stable, I could include it in > the main Speex distribution. Hmmm, define working and stable :) <braindump topic="speexcat"> It began as a merge between speexdec and speexenc from 1.0beta3, with the encoding/decoding removed, and simply piped in and out from ogg

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

Hello, > 2. I don't have a good source of wav data for testing. I've noticed that > introducing bugs into speex (even gross ones like returning completely > incorrect codebook entries) tends to, sometimes subtly, degrade quality > instead of blowing up. Is there an existing set of source data - and > maybe even a test harness that will do binary comparison, so I can avoid

Hi, Speex is getting close to beta4, which I'd like to be feature-complete (or as close as possible). That's why I'd like to ask if anyone here has needs for a feature that hasn't been implemented yet. If so, please let me know. For those interested, here's what's going to be in beta4. First, the VBR code has been greatly improved and now works good with wideband too.

On Tue, Dec 17, 2002 at 11:55:21PM -0800, Greg Herlein wrote: > If such a thing happens, discussion of the RTP profile draft > would be most welcome - please get responses back to the > list! Now, if this were finalised before the conference then we could do a demo and use it for broadcasting the lectures streams around the world... What is currently the best way of doing this? I'm

Hi. I'm wondering if anyone out there can give me a little help trying to figure something out: I presently am running CentOS4.1 with GNOME as my desktop (It runs quite well, I must say!). I just have a little curiosity: I have a few audio files (.wav, .mp3 and .ogg) and I like to "preview" them. When run my mouse over the .wav files (and mp3 files) the

Hi, Can someone please comment about the below issue [root at host0040 kaushal]# file obd-demo.mp3 obd-demo.mp3: MPEG ADTS, layer III, v1, 256 kBits, 44.1 kHz, Monaural [root at host0040 kaushal]# sox obd-demo.mp3 -e stat sox: Failed reading obd-demo.mp3: Do not understand format type: mp3 [root at host0040 kaushal]# sox -V obd-demo.mp3 -r 8000 -c 1 -t ul -w vm-intro.ulaw sox: Failed reading

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'm a new Asterisk user. I've managed to set it up to do everything I want except sound good. Currently, Asterisk sounds considerably worse than my cell phone. I know VOIP can be _better_ than my cell phone, because I've heard Skype do it. (Using 32k iLBC, I believe.) I did an experiment with audio quality: 1) I made a recording which was pretty good. I used an iSight

On Sat, Dec 14, 2002 at 06:04:08PM -0500, Jean-Marc Valin wrote: > Maybe not everything got re-compiled and some parts were left with -g or > something? With the files you sent, I get (on my PIII 1 GHz, with -O3): > speexenc --comp 3 --quality 3 sample.wav sample.spx > 0,65s user 0,00s system 100% cpu 0,647 total > > strange... I'm recompiling the lot, no debugging, -O3 (or

Hi, Can someone please comment about the below issue [root at host0040 kaushal]# file obd-demo.mp3 obd-demo.mp3: MPEG ADTS, layer III, v1, 256 kBits, 44.1 kHz, Monaural [root at host0040 kaushal]# sox obd-demo.mp3 -e stat sox: Failed reading obd-demo.mp3: Do not understand format type: mp3 [root at host0040 kaushal]# sox -V obd-demo.mp3 -r 8000 -c 1 -t ul -w vm-intro.ulaw sox: Failed reading

For my MP3 files on my FreeBSD box, I can type mpg123 to play them. For WAV files, I can use wavplay, but it can't take stdin. Does FLAC have a simple command-line player for Linux/BSD anywhere?

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,

Don't know if anybody is still missing the lame oggenc features for resampling, lowpass/highpass filters etc, but I wrote a little script that uses sox to do all the stuff I need to real-time encode oggenc from the radio, or any input device. #!/bin/bash DATE=`date '+%m-%d-%Y-(%H.%M)'` DESTIN=/video/music/perftoday export DATE=$DATE'-PerformanceToday.ogg' sox -V -r 44100 -c