similar to: echo cancellation (power19550114fover)

Narrowed it down a bit. For me the code crashes encoding the second frame if the first encoded frame was all zeros. Can someone else reproduce this or am I on crack and should look for a bug elsewhere? // Initialize speex encoder. SpeexEncoderBits = new SpeexBits; SpeexEncoderState = speex_encoder_init(&speex_nb_mode); INT FrameSize, SamplingRate = 8000, // 8000 Hz BitRate

Hi All, I am trying to use Speex + PyAudio to convert voice to text using Google Speech API. I downloaded PyAudio Source, PortAudio and Speex and installed portaudio, followed by pyaudio with static link and then speex also I installed. I am able to import both speex and pyaudio into python fine. But when I call these functions: e = speex.Encoder() e.initialize(speex.SPEEX_MODEID_UWB) I get a

Hello, I am JungAh Lee of Nuance communications. I have a question about the license part described in http://www.speex.org/fsos/. Does it mean I don't need to provide the application source code, though I modify the speex source code or use a part of it within my program? Regards, JungAh -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am seeing some problems when using the Powerware 5115 UPS when connected to the Patsburg USB controller and would like to know if anyone has a solution. The symptom is that PING messages sent to the bcmxcp_usb driver take as long as 25 seconds to complete. The configuration is a Romley patform (a.k.a., Sandy Bridge + Patsburg) running Debian 6 with 2.6.39 kernel with NUT 2.6.4 in which the

Hi, I am an user of speex codec.Can I ask some question about the speexcodec 1.2beta2 on installed to linux(fedora 6) thanks! After I downloaded the latest version of codec then I type the commands to install : configure -prefix=/home/...../test -enable-shared -enable-static make make install then I try to compile my encode.c here is the encode.c of mine: =====================================

Hi,I'm having performance issues encoding with speex on an iphone4 + iOS5. The same code yields excellent results on iPad2 + iOS5, Mac OS X 10.6 as well as Windows 7. The call to speex_encode_int is takes 0.0065 to complete. Because encoding is so latent the receiver incurs underruns. Here is my speex setup: ? ? unsigned val = 0; ? ? /**? ? ?* Initialize the encoder in narrow band mode.? ?

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

Hello, in my project im using speex 1.2rc1 and the preprocessor VAD seems to only separate complete silence from not complete silence frames. The Speex Manual, you can read "The voice activity detector (VAD) provided by the preprocessor is more advanced than the one directly provided in the codec." but if you go to the source code in preprocess.c line 995 "/* FIXME: This VAD

Jesus, Unfortunately, FFT and magic algorithms don't work (yet?). You might want to try this if you're not satisfied with Speex VAD: http://lists.xiph.org/pipermail/speex-dev/2008-August/006860.html It won't perform any miracles, but I think it works pretty well and is easy to tweak. Tom >---- Original Message ---- >From: jmorion at toomeeting.com >To: speex-dev at

Thanks for the advice. With complexity=1, bit rates 5950,8000 and 15000 (CBR only), I'm still getting 30-50 seconds encoding time for a 10-second file. To anyone who has made this work in ARMv4, or knows how to, can I get some advice on the settings? FIXED_POINT is already defined in all relevant files. My source code is patterned after sampleenc and I haven't tried using Ogg.. my source

Hi Alls Sorry for my newbe question I want to decode a speex multi frame stream I got 38 bytes packets and make 800 bytes buffer this is my simple code CSpeexCodec::CSpeexCodec(void) { // speex_bits_init(&m_speexBitsOut); m_pSpeexEncStateOut = speex_encoder_init(&speex_nb_mode); int nEhc=1; int nSamplePerSec = SAMPLES_PER_SEC; static int nQuality = 8;

Hey All I''m Using Xen 3.0.3-105 on CentOS 5.5. It Has to Nics: eth0 - Internet eth1 - Internal Lan ( where Dhcp , DNs , Cobbler - Kick Start Server Resides ) I''ve Set xend-config.xsp to create xenbr1 on eth1 and it looks well So Guest May Get Access TO ALl Resources via the Bridge. brctl show bridge name bridge id STP enabled interfaces xenbr1

Hi, I have a module sampling raw PCM data on Win CE as 10ms time slice (160 bytes), mono, 8000HZ, 16 bits per sample. Does anyone know what is the mflops for using fixed point on a Win CE compared to using floating point? Looking at the manual, "In practice, frame_size will correspond to 20 ms when using 8, 16, or 32 kHz sampling rate." for a 8 kHz sampling, the framesize should be

Hi, in my application i quite often (not allways) get those two warning: "Invalid mode encountered: corrupted stream?" "Invalid wideband mode encountered: corrupted stream?" The stream is not corrupted but i wish to know what those warning mean to correct my code. My encode and decode routines are those (libspeex 1.1.4): SpeexBits enc_bits, dec_bits; void *enc_state,

hy, recording and playback without speex is working perfectly Lp, Tim +--------------------------+ | email: rico at gama.us | | www: http://gama.us | |--------------------------| | tel: 00386 31 457 627 | +--------------------------+ 2009/9/2 Tim Rijavec <rico at gama.us> > hy, > > here is my speex encoder/decoder .. the sampleRate i use is 16000 and >

hy, recording and playback is working perfectly without speex. i have try to set samplerat from 6000 to 441000 and quality from 1 to 10 sam with complexy, but the best i can get is with 16000 samplerate, 5quality and 3complexy .. but still, the voice that came out is annoying, artificial, robot ,... Lp, Tim +--------------------------+ | email: rico at gama.us | | www: http://gama.us

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, 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 );

-------Original Message------- From: Eid Date: 10/09/05 14:07:44 To: speex-dev@xiph.org Subject: problem in encoder/decoder hi all, I'm developing an app to broadcast voice over web. i tried to use speex and download ver 1.04 documentation and use it to develop encoding and decoding but after decoding voice is damaged i attach my encode/decode function. BOOL

Hi, I'm using the latest version of libspeex (checked out today) and have some problems with checking the target bitrate in the encoder. Calling: peex_encoder_ctl(globals->enc_state,SPEEX_GET_BITRATE, &bitrate); Seems to set bitrate to 24600 no matter what the qualitysetting is.. The functions used previous to this call is: peex_bits_init(&globals->bits);