similar to: reduction of noise due to high microphone gain

This works really well for white noise reduction. However what I've noticed was the amplitudes of normal speech samples also get reduced. Is this something by design, or is there a way to automatically recover the original speech sample volumes ? <p>Thanks. <p>Tongbiao <p>-----Original Message----- From: Jean-Marc Valin [mailto:jean-marc.valin@hermes.usherb.ca] Sent:

Le dim 31/08/2003 à 20:12, Daniel Vogel a écrit : > > This works really well for white noise reduction. However > > what I've noticed was the amplitudes of normal speech samples > > also get reduced. > > Noticed this as well recently. This is probably due to the AGC (Adaptive Gain Control) that's integrated with the denoiser. I'll try adding an option to

I tried feeding in the 3 encoded frame in ONE BLOCK, and calling speex_decode() 3 times in a roll. Only the 1st frames came out perfectly. For the other 2, I got "corrupt" frame warning. I was supposed to get 38 bytes consumed each frame (narrow-band, VBR off). I tried speex_bits_remaining() to peek on the # of bits consumed, and got variable (clearly wrong)#s returned. But if I

Hi all again, Thanks for the response to my previous question, Jean-Marc. It was of great help. Now another problem I encountered: How do I detect that comfort noise is being sent out (I have VAD, DTX on)? I could not find an API function for it, so tried frame rate. For nb, it gives 2950-3950 for silence, however it seems to also give low number sometimes even for speech periods. I

Hi, I am considering to use Speex in a particular environment and I want to ask if you thought it realistic. I am going to use Wireless SIP (G.711) handsets in an environment that is typcially noisey (background), connecting to an Asterisk PBX. I am considering converting the G.711 to Speex so I can "massage" out the background noise. Is this realistic? Any input would be great.

Understand that it's a *guess*. If this fixes or at least betters the situation, you're going to need to find a legit way to insulate these cards, or to switch cards. If it's for customers, I should hope you wouldn't be using enamelled aluminum foil. :D -----Original Message----- From: Tongbiao Li [mailto:tli@viack.com] Sent: Thursday, September 18, 2003 11:20 AM To:

I'm not sure you'd have much success in trying to make a generic profile to fit the "typical noisy PC" since PC configurations vary so much. Consider an integrated AC97 codec vs. the mic input on an SB Live!/Audigy Live!Drive, or whether or not the input from the CD-ROM audio is muted, or what noisy devices might be situated near the mic. It's nice to be able to get

Hi, I'm not sure how speex_denoise() works, but my solution to this problem is fairly straightforward and works well in practice (in a simple VoIP program.) Note that I am also an amateur and am making up some terminology here... but I've tested my approach on several noisy environments with low-quality mics and it does work well. Problem: There's usually annoying noise present

Take what I say with a grain of salt: I'm an amateur and haven't actually touched Speex in any way, yet. I'm just sort of passing on personal belief from personal experience. Also, check and make sure that the microphone line is insulated. There are a number of problems with sound cards picking up interference from the host machine. The wires that run between ICs on a card

Thanks for the speedy response and detailed, enlightening explanation. Now I understand where the problem is, and will try out your suggestions just to further confirm my conjecture. When I am done, I have to take the foil out, though. This is a product for our customers to use, and although we've got budget for mulffing every sound card we developers use, most likely the company

That's might be speex_denoise() is trying to do already, based on the fact that if noise is present in speech signal, the former gets suppressed to a certain extend. Also, I noticed after suppression, the short, noisy attack - speex_denoise() have to be doing some kind of profiling. I've looked at a few offline denoise() solutions. They all involve profile building. A real-time

Hello, We currently have code for sampling at 8 KHz at device layer. In order to have minimal code change while still doing wideband I am thinking of packing two 20 ms frames of the low band data as input to Wideband mode for encoding. Is this a feasible idea? I actually tried it out. Speech came out at the decoding end, but I could occasionally hear low pitched, high frequency

Hi, I'm trying to use a denoiser on a wince with a FIXED_POINT defined. Denoiser works OK - it removes the noise, but then it unacceptable hurts a voice. Here is a code that I use: #define TEST_DENOISE_SAMPLES 2000 void test_denoise() { FILE *fin; FILE *fout; spx_int32_t rate=0; int chan=1; int fmt=16; int denoise_enabled = 1; SpeexPreprocessState *preprocess;

Here's the situation: 3 frames are being transferred back to back over RTP. The total payload is known (thus the TOTAL frame size). We'd like to decode all 3. According to IETF RTP Profile (appendix C, of Jean-Marc's Speex Codec Manual, verison 1.0), "it's only necessary to consider the sequence of bits as output by the Speex encoder, and present the same sequence to

Hi Richard, Richard McGravie a ?crit : > We are looking for a consultant to assistance in the following areas. > 1. Shim in the pre filter from speex to all trunk interfaces in > asterisk. (E1, ISDN and FXO) > > 2. Optimize and improve the pre-filter for airplane noise floor. > Is this something you would be willing to assist us with or do you have > any contacts that could

Hi all, i use speex preprocessor features in this way: =================================== #define NN 160 /* 20msec di audio */ ... int tbc=0,c,d,ret; spx_int16_t TEMP_Buffer[NN]; char DLECODE; /* Inizializza il preprocessore Speex se non inizializzato */ if(Modem->speex_pp_state == NULL) { Modem->speex_pp_state = speex_preprocess_state_init(NN,AUDIO_SAMPLERATE); }

Hi! I'm using the newest (beta2) speex library on PocketPC (Windows powered) and experiencing some problems with noise suppression preprocessor turned on while encoding 44kHz files. You can hear the example here: http://szalik.net/speex (this is a 44kHz, 16, mono file encoded in UWB mode) I tried using speex cross-compiled with gcc (fixed point + arm4 asm) and VS (just fixed point) and it

I have (finally) posted my Speex wrapper classes. They are at: http://www.grandgent.com/spx/tgAudioCodec.zip I followed your recommendations and they worked fine with 1.1.0. However, I'm still having the same problem with 1.1.7 that I had the last time I tried to upgrade. I'm using the same code with both versions, except for calling speex_encode_int instead of speex_encode, and

I'm getting confused, what's the problem exactly? The AGC doesn't remove the DC or it adds one? Jean-Marc Aymeric Moizard a ?crit : > Hi Jean-Marc & all, > > I'm currently working on a wav file with very low volume. > This wav also contains a DC component. > > Using testdenoise with modification (AGC turned on with > 20000.0 value) I'm able to

Hi everyone, Our team is working on a realtime voice communication application. We are using openAL 1.1 to capture the samples and then encode them using speex. On the remote machine we decode the samples and play them using openAL again. Capture and play formats for the samples is AL_FORMAT_MONO16 and frequency is 22050. we are using wide band encoding/decoding. The encoding sample