similar to: frame of silence

Okay, you've answered part of my question, which is "What value equals silence?". I assume then that a (decoded) frame of silence would be a buffer the size of my frame (320 bytes) full of 0's. Passing this frame (a frame of all 0's) through the encoder causes it to blowup though.. In response to your answer below, I don't think I want to overwrite the decoded

Hi All, I am getting this error during make. please help me./ speexec.c: In function `speex_ec_process': speexec.c:112: syntax error before "noise" cc1: warnings being treated as errors speexec.c:133: warning: implicit declaration of function `speex_echo_state_reset' speexec.c:148: warning: passing arg 5 of `speex_echo_cancel' makes pointer from integer without a cast

In what fashion does the encoder 'blowup' when passed a frame of zeroes to encode? What version of speex are you using? ________________________________ From: Chris Cowden [mailto:ccowden@ncsoft.com] Sent: Thursday, 15 February 2007 7:57 AM To: David Hogan; speex-dev@xiph.org Subject: RE: [Speex-dev] frame of silence Okay, you've answered part of my question, which is

Forgive me if I misunderstood your question, but can't you just overwrite the decoded audio buffer (either before encoding or after decoding, depending on what you are trying to do) with zeros? Cheers, David Hogan ________________________________ From: speex-dev-bounces@xiph.org [mailto:speex-dev-bounces@xiph.org] On Behalf Of Chris Cowden Sent: Thursday, 15 February 2007 3:43 AM To:

But that's what a jitter buffer will do, eventually - drop a packet. And it's better to drop a silence packet. Cheers, .a -----Original Message----- From: Jean-Marc Valin [mailto:jean-marc.valin@usherbrooke.ca] Sent: Wednesday, December 21, 2005 3:39 PM To: Alex Bakaev Cc: Shantanu Gramsci; speex-dev@xiph.org Subject: RE: [Speex-dev] Bitrate problem On Wed, 2005-12-21 at 15:29 -0800,

Hi there just installed latest build of wine through macports. I'm running MacOS 10.5, with intel core 2 duo 2.4GHz. When I click the audio tab in winecfg it gives me the message: There is no audio driver currently specified in the registry. A recommended driver has been selected for you. You can use this driver or select another driver if available. You must click Apply for the selection

Sounds like either your machine is too slow or you have some other buffer issue. A decode thread is usually ideal so that data can be decoded before it is needed by the buffer fill operation. You might be able to get a sasifactory solution by increasing the number of buffers and using them in a round-robin fashion 1-2-3-4-1-2-3-4. Wouldn't do any more than 1/4 sec per buffer. Cheers, Chris

I am having trouble running TalonRO which is a ragnarok online private server. I am on Mac OSX 10.6.7 Snow Leopard and I am using Wine 1.3.16. I downloaded the installer file for the game and ran it through wine. The installer seemed to run fine without any problems. After that I ran the patch client for the game which then let me boot into the game. After that the game boots up but it just

well i discoverd, HL uses waveIn, which is a part of the DirectSound ... so ... is there a Wine binding for waveIN ? - -moritz angermann

>I remember?I had to expose the echo cancelation level implementing a get_echo_level( ) function based on this: >http://lists.xiph.org/pipermail/speex-dev/2008-September/006889.html This is really a good idea to determine the frequency difference between capture and play of the sound card. But it need constant far-end voice and a long time because it must repeat the process of

[Just curious, and seizing the opportunity to communicate with other folks who are doing the same kind of thing I am...] How are you measuring the latency? I tried measuring it with my program (also Win32-based, also using DirectSound[Capture]) and came up with around 130ms. To measure it, I placed the mic near a speaker to get feedback going, had my program connect to itself (local

Hi, I'm working on making a game in VB2008, and I'm hoping to use OGG Vorbis-format music (because ogg files are tiny =D) in it. I've googled for info on doing this, but all I've found are dead links and non-free implementations... and when I tried compiling libogg ages ago on Windows, I completely failed. I've already worked out how to use DirectSound, so I really just need

I'm having a problem trying to write a simple vorbis file player using DirectSound. The decoding portion is pretty much straight from the vorbisfile sample code. The pcm data gets put into one half of a DirectSound buffer, and as that half is playing, the next half gets filled. The problem is, there's skipping and some noise when the file is being played (although some of the music is

Hello, On the Windows platform, I would like to compile a version of the codec files ogg.dll, vorbis.dll, and vorbisfile.dll that are small. I notice that the provided Windows projects are very large...the release Vorbis.dll is > 1 mb. To me, that is huge. I am only using decoding features. I looked in the Zinf project, and they seem to have a version with the encoding functions

First of all, thank you for your input Tim. That is very helpful. I would love to hear from other people with experience of AEC and Speex. I guess I have to split my question into to parts now. 1. Is it a fact that using the windows multimedia API (wave audio) for audio capture and playback makes it impossible to do echo cancellation with Speex AEC or other EC method due to inprecise timing? I

This is a common problem. Sound cards run at different clock speeds (but your difference seems kinda high). What I'd suggest is dropping silence packets once in a while. .a -----Original Message----- From: speex-dev-bounces@xiph.org [mailto:speex-dev-bounces@xiph.org] On Behalf Of Shantanu Gramsci Sent: Wednesday, December 21, 2005 2:05 PM To: speex-dev@xiph.org Subject: [Speex-dev] Bitrate

Hi All, In order to deal with acoustic echo cancellation problems of most PCs which sound cards have different capture and play frequencies. I made a trial. At first, a 1000Hz sine wave is played for a long time via a speaker and its acoustic echo is recoreded. Seconds, get the frequency of the echo by a FFT analyser. So the difference between capture and play frequencies is obtained. Thirdly,

Running OS X 10.6.3 on an Intel Mac (processors "2.33 GHz Intel Core 2 Duo", memory "3 GB 667 MHz DDR2 SDRAM) and the newest version of Wine accessible through MacPorts. Hey. A little while ago, I was having trouble installing wine. I eventually managed to install with MacPorts. Everything was going fine until today, when the program I actually wanted to run (Dwarf Fortress 0.31.01)

Code: Last login: Sun Jul 26 02:37:47 on ttys000 christopher-burkes-macbook-pro:~ cpburke$ wine /Users/cpburke/.wine/drive_c/Program\ Files/Continuum/Continuum.exe ***err:d3d_caps:IWineD3DImpl_FillGLCaps Invalid nVidia version string: "2.0 NVIDIA-1.5.44"*** fixme:win:EnumDisplayDevicesW ((null),0,0x32f2fc,0x00000000), stub! fixme:winsock:WSACancelAsyncRequest (0xdeba),stub

Thank you. But it will cost you a long time to get the accurate play and capture frequencies. Does your program test two frequencies of the sound card each time? Because different sound cards have different frequency errors. And the resampling program is also time consuming because the target frequency is so close to the sampling frequency of the input signal, isn't it? I have tested program