similar to: configuring speex for narrow band

Hello, I am trying to use speex for speech encoding on my ARM based device. I am facing issues with quality of voice with the narrow band configuration (8k sampling rate). With complexity set to 3 and VBR enabled, I am getting high bitrate output. As an example, My input is 8K stereo and I am passing the encoder with 160 samples at a time as per the manual and samples. When there

Hi Marc, I've downloaded the Speex1.2Beta version. In the code still there are some float variables are used for the Narrow band decoding. In the file "nb_celp.c" .93 is getting multiplied with the GAMMA_SCALING [Line number From - 1341 - 1351] Can please tell me weather the 1.2Beta version has complete narrow band decoding in fixed point?? Thanking you, Regards, T

Hello, I would like to run a script over a speex file which determines, how long the speech contained in the speex file is. So I run through the ogg file, skip the first two ogg _pages_ (speex header and comment header, both contain IMHO only 1 ogg _packet_), and then count in any ogg page the number of the ogg _frames_ (which, unless I used -nframes while encoding, which I did not), and multiply

I can only evaluate this with my subjective point of view. I had a special test scenario doing chat with cheap webcam microphones and loudspeakers. Fraunhofers solution was the only one that could eliminate the echo. In double talk the quality gets lower but is still very good. You might want to ask Fraunhofer for a demo version to test for yourself. I have no details on the algorithms being

Thank you, Andreas Engel. I downloaded the white paper of the Fraunhofer Acoustic Echo Control. http://www.iis.fraunhofer.de/bf/amm/download/whitepapers/Acoustic_Echo_Control-wp.pdf It said > "In the Fraunhofer Acoustic Echo Control, the frequency spectrum of the microphone signal is > modified so that the undesired echo components are removed from the signal transmitted to > the

Hello guys, I get the following error whenever i try to check if the following pass exists. I am using Mac OSX leopard and gcc4 opt -load ../Release/lib/LLVMHello.dylib -hello Error opening '../Release/lib/LLVMHello.dylib': dlopen(../Release/lib/LLVMHello.dylib, 1): Symbol not found: __ZN4llvm12FunctionPass11runOnModuleERNS_6ModuleE Referenced from:

On Tue, Jan 17, 2006 at 02:19:17PM -0500, me@sbooth.org wrote: > Hi all, > > I have recently added Speex support to my app (http://sbooth.org/Max/) and > am in the process of creating a GUI for the user to control the codec > parametere. I am new to Speex and as such I am not really sure which > parameters are more important than others, and deserve prominent > placement,

Hi, There is reported bug about this (#2771), or at least related. http://llvm.org/bugs/show_bug.cgi?id=2771 Though it seems there is no "proper" solution known so far. The workaround is to use 'opt' from the Release/bin directory In your case: ../Release/bin/opt -load ../Release/lib/LLVMHello.dylib -hello Would probably work. Best regards Vlado > Hello guys, > >

hi, I have a small and quick question. When speex divide the UWB speech signal into High and low bands, JMV said to me in HA that : "For ultra-wideband (0-16 kHz, 32 kHz sampling), I first split the band into wideband (0-8 kHz) and "very high band" (8-16 kHz). Then, the wideband itself is split into low (0-4 kHz) and high (4-8 kHz) band. So there's a total of 3 bands

Than a lot John!!!! That was great, i wanted to read the performance counter, but was always afraid how llvm would react (virtual machine) Thanx for the resource i ll go through it. i think i read somewhere that llvm doesnot support assembly instructions??? Also can you redirect me to run-time transforms for llvm, where can i find the material. I thought that the transfor pass is done on the IR

Hi everyone, I have been using vorbis for ages as audio codec for archiving movies and I have always been happy with it, but now I have been thinking about two things: I) Multi-channel Up until now I mostly encoded to stereo; now I am thinking of encoding more movies to 4.0 or 5.1. I heard that vorbis does not optimize for channels > 2, i.e. that 6 channels is just stereo*3. AAC is

Hi everyone, I have been using vorbis for ages as audio codec for archiving movies and I have always been happy with it, but now I have been thinking about two things: I) Multi-channel Up until now I mostly encoded to stereo; now I am thinking of encoding more movies to 4.0 or 5.1. I heard that vorbis does not optimize for channels > 2, i.e. that 6 channels is just stereo*3. AAC is

Hi everyone, I have been using vorbis for ages as audio codec for archiving movies and I have always been happy with it, but now I have been thinking about two things: I) Multi-channel Up until now I mostly encoded to stereo; now I am thinking of encoding more movies to 4.0 or 5.1. I heard that vorbis does not optimize for channels > 2, i.e. that 6 channels is just stereo*3. AAC is

Hi, JCCTray is a Java port of CCTray that was originally written in .NET. JCCTray is a utility for use with all flavors of CruiseControl Continuous Integration servers. Specifically it works with CruiseControl, CruiseControl.NET, CruiseControl.Rb. It provides feedback on build progress, and allows control over some of the server''s operations. You can download JCCTray from

Hey, We hit a bug today because Arel::Visitors::ToSql is not threadsafe. Here is what is happening: Arel::Visitors::ENGINE_VISITORS is a cache of visitors instances. These instances are not inherently threadsafe because it contains state ''@last_column'', ''@connection'' that is shared between threads. The other variables ''@pool'',

Ok here is something I couldn't see on the description page. We need to convert to 8 bit audio right? I'm assuming narrow band is 8 KHz 8 bit. Wide band is 16 KHz 8 bit. UW band is 32 KHz 8 bit audio? George -----Original Message----- From: Tom Grandgent [mailto:tgrand@canvaslink.com] Sent: Tuesday, October 03, 2006 11:42 AM To: Bertie Coopersmith; George Ou Cc: speex-dev@xiph.org

Take a look at this: peexenc --vbr -V --quality 4 audiodump.wav a.spx Encoding 32000 Hz audio using ultra-wideband (sub-band CELP) mode (stereo) Bitrate is use: 22400 bps (average -1268444817 bps) I'll take a look at what the problem is later today or tomorrow, but if anyone beats me to the bug, let me know before I spend too much cscope time playing with it. Regards, Matthias --

> 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 students (non-students may skip this)! As some of you may be aware, the deadline for the Google Summer of Code (http://code.google.com/soc/) program is approaching quickly (March 24th). For those interested, there are two Speex-related projects (below) and lots more in Xiph.Org in general (see http://wiki.xiph.org/index.php/Summer_of_Code). You can apply at

Greetings list, I am working on a project on which we wish to use Speex with Google Automatic Speech Recognition (ASR) to transcribe Speex audio being sent on to Google ASR service and return us the text of the spoken audio in the Speex audio stream. However, Google ASR's Speex support requires the off-standard Speex-with-header-byte format, and my group cannot find any worthwhile