similar to: trying to use latest but can't link now

Hi, I use the pre-compiled libs of speex, version beta3. I've got this link problem when I use the fonction speex_encoder_init(&speex_wb_mode) error LNK2001: unresolved external symbol _speex_wb_mode I made the link with the librairies libspeex.lib and libspeexdsp.lib in my properties' project. I used the dumpbin executable of VC to see the external variable defined in the libs,

Hi, No, it only happens with speex_wb_mode. But like I said, it is because in wideband mode, the function sb_encoder_init is called and this function initializes the sampling rate using speex_*de*coder_ctl instead of speex_*en*coder_ctl In narrowband mode, the sampling rate is just initialized to 8000 st->sampling_rate=8000; BTW, I saw that my email was added to an existing thread instead of

Hi, I'm using speex 1.1.4 and when I query SPEEX_GET_SAMPLING_RATE as in speex_encoder_ctl ( m_pEncoderState, SPEEX_GET_SAMPLING_RATE, &m_SampleRate ); I get 2 The encoder is initialized with m_pEncoderState = speex_encoder_init ( &speex_wb_mode ); On the decoder side, the correct sampling rate is returned: m_pDecoderState = speex_decoder_init ( &speex_wb_mode );

On 12/13/07, Jean-Marc Valin <jean-marc.valin@usherbrooke.ca> wrote: > > I am a newbie here and just started using speex in our lab. > > During compiling my voice application with windows-compiled version provided > > today, I am getting "unresolved external symbol _speex_wb_mode". > > I am probably missing some setting with VS6.0 linker... (I am a bit new

Hi All, I'm very new to speex and in fact handling audio at all, it seems I have run in to a problem I seem unable to fix. I'm trying to take audio from a microphone using alsa, then encode it as speex and save to disk. I have been wondering if it has something to do with endian type, but speexenc and speexdec works fine. Currently I have the following setup: Platform:

Hi there. I am having a little trouble understanding the frame sizes chosen by the codec. testenc_uwb.c from the speex-1.0 source distribution has a framesize of 640 hardcoded and makes use of this value exclusively. However, a mode query on the actual codec returns 320 as a framesize for this mode. int tmp; speex_mode_query(&speex_uwb_mode, SPEEX_MODE_FRAME_SIZE, &tmp);

I am using speex for encoding and decoding speech files in a speech synthesis program. I am using concatenative approach for speech synthesis using diphones (small speech chunks) as basic unit. Now in my application, the order of diphones to be joined after decoding is determined at runtime. Therefore I have to encode each diphone independent of others. I find out that when I encode diphones

Hi Tom, Thanks for your reply, I'm having a few difficulty's following you advise. You mention that I should be calling 'speex_encoder_int' & 'speex_decoder_int', I'm having problems finding these functions in the speex header. I am currenlty calling speex_encoder_init & speex_decoder_init in my test code, I have repeated the relevent bits of the example

Hi all, I am a newbie here and just started using speex in our lab. During compiling my voice application with windows-compiled version provided today, I am getting "unresolved external symbol _speex_wb_mode". I am probably missing some setting with VS6.0 linker... (I am a bit new to windows too) If anyone could point it out, I would appreciate it. - Tim -----Original Message-----

Nicer way: void* speexState = speex_encoder_init(&speex_wb_mode); int speexFrameSize, speexRate; speex_encoder_ctl(speexState, SPEEX_GET_FRAME_SIZE, &speexFrameSize); speex_encoder_ctl(speexState, SPEEX_GET_SAMPLING_RATE, &speexRate); SpeexPreprocessState* speexPreprocessState = speex_preprocess_state_init(speexFrameSize, speexRate); Jozsef -----Original Message----- From: Max

In short try calling speex_encode_int and speex_decode_int. The calls to speex_encode and speex_decode expects the data to be a floating point values. The data is passed by pointer and the compiler does not do the conversion. This is why you are getting the segfault on decode as the short is 16-bits and a standard float is 32-bits. Tom -----Original Message----- From:

Hi all, I have a problem when I am encoding (or decoding) stereo audio. With mono data, things are fine and everything works without any problems. When I try to decode stereo data, all I get is a static sound - similar to that of a radio not tuned to any specific station. I wonder what might be wrong? Below is the code, first, of the encoder and next that of the decoder. Any information or

hi all, i'm new to speex and i'm stuck on a problem. I have a buffer that holds 5 seconds of audio data, sampled at 16000 hz and quantitized with 16 bits pcm. I only want to encode it with speex and then decode it and write it back in the same buffer. no compiling errors and no errors during runtime, but from the sample of 5 seconds only a short noise is all I get. I hope you have any

As for me - speex_echo_cancellation is a better choise. Try using it in capture thread instead of those speex_echo_capture and speex_echo_playback functions. And please, describe your problem in details. Cause the fact that you "didn get echo cancellation" doesn't mean you are doing smth wrong. Regards, Anton A. Shpakovsky -----Original Message----- From: speex-dev-bounces at

Hi all, I'm new to speex and there are quite a number of things I am stuggling with. Any help much appreciated. -bob- Environment: Ubuntu 10.04 Speex 1.2 rc-1 GNU C compiler IDE Code::Blocks My application is actually 'D' but I've written a 'C' wrapper following the examples on speex.org. I have raw data which is in float format with a nominal block size of 512 samples

hello, we start to develop an open source project. This project is a VoIp application with the same features like "Teamspeak" (http://www.goteamspeak.com). Our test application run in peer to peer mode under windows 2000 and above. At the moment we are using the gsm codec (13 kbps bandwidth). Later there will be a port for *unix systems. As aforementioned the project will be released

Anton A. Shpakovsky <saa <at> tomsksoft.com> writes: > > As for me - speex_echo_cancellation is a better choise. Try using it in > capture thread instead > of those speex_echo_capture and speex_echo_playback functions. > > And please, describe your problem in details. Cause the fact that you > "didn get echo cancellation" > doesn't mean you are

hi there, I am new to mailing list so excuse me if I don't obey to the 'netiquette'. i am writing voice chat and speex is in the root of it. i write it in Java and use JNI to link with 'C'-based Speex 1.2beta. [I know of JSpeex but there are not implemented some features] recently i decided to use DTX feature of speex as well. the code follows. The problem is that no matter

Anton Korobeynikov wrote: > Hello, Jonathan > > >> I thought I'd messed up something related to the linker, >> but I couldn't explain why XP would work using the same steps. >> > The only thing comes to my mind seeing this: perl from msysDTK works > differently (somehow) on Vista, thus llvm-config is broken which leads > to missed libraries,

It seems that speex (speex 1.0.4 on OSX 10.3.9) is writing over the input buffer (with what looks like a sine wave). Included below is a short program that demonstrates the problem. Is this a bug, or am I doing something wrong? I don't see this behavior mentioned in the docs. I don't really need the input buffer, but I'm guessing this problem is related to a high pitched sound