similar to: Computing pcm size

Hello, I'm trying to use the speexlib but I just can't do it. I've made just like the example (on the site) says. Some times I get SIGFAULT. Also, I'm reading the input_frames from /dev/dsp (directly). What is missing here? speex_bits_init(&bits); enc_state = speex_encoder_init(&speex_nb_mode); speex_encoder_ctl(enc_state, SPEEX_GET_FRAME_SIZE, &frame_size);

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

When I just copy the microphone input buffer to the output buffer the sound plays OK. But if I encode and decode the buffer through Speex I get a high pitched constant tone in the background. I actually do hear my voice speaking when I talk, but it's faint and much quieter than the tone. Here's what my data looks like: Input is the first 5 floats of each input buffer. Output is

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

Is there any particular reason why ov_read() packs floats to integer PCM inline, rather than being implemented in terms of ov_read_float() and a separate packing fucntion? There are obviously many advantages doing audio manipulation on the floats before packing, but right now you have to reinvent the packing stage yourself - in a replaygain backend that I'm working on, I ended up copying

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

Hi All, i need your help in determining the problem in the following sample code (taken from speex manual) for fixed point encoding. i tested encoding a 160 sample frame and then decoding it back But The problem is that the decoded frame is totally different from the original frame(see output below).What is the possible reason for this?.thanks for any help #include<stdio.h>

Hi all Sorry if this is a dumb question: does the input_frame passed to speex_encode_int *have* to be frame_size samples long? e.g., If I only have 100 samples left to encode (which is less than the frame_size of 160 samples), can I just use an array that contains 100 samples, or do I need to create an array containing the 100 "real" samples plus 60 null samples at the end?

Changing from using floats to shorts did fix the high pitched tone problem. I'm having other problems but I'll look into it more first. SteveK wrote: > > On May 21, 2006, at 6:33 PM, Kevin Jenkins wrote: > >> When I just copy the microphone input buffer to the output buffer the >> sound plays OK. But if I encode and decode the buffer through Speex I >>

This is a proposal for Ogg MIDI streams. It will also help other stream additions to Vorbis, as they should follow a similar model. Ralph has done much of the work on this, and also is working on MNG Ogg streams. Background ---------- I was contracted to do this work, so even if you can't think of a use for this, there is one ;) Basically with midi in Ogg, we can have a synchronized

Hi i'm developing a sort of VoIP application for my ipaq using speex... I'm still at beginning and i have many problems encoding and decoding my wav files....output is only noise! Why? I'm using Libspeex 1.1.3, Embedded VisualC++ 3.0, Ipaq 3850(206 MHz Intel® Strong ARM 32-bit RISC Processor) PocketPC 2002 (Windows CE 3.0). Libspeex is complied with the definition of

hy, here is my speex encoder/decoder .. the sampleRate i use is 16000 and quality,complexity are at 5. can someone take a look in to the code and see if there is something that is making that robot voice here is a link to colored and numbered code, same as below http://barvanjekode.gama.us/temp/1257361243.html thanx. code ----------------------- #include "codec.h"

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

Il 14/04/2010 14:37, Randy Yates wrote: > > Usually a buffer is one frame of data, and a frame is 20 milliseconds. > Since the sample rate is typically 8 kHz in narrowband mode, this > corresponds to a buffer size of 160 samples. Hi Randy, thanks for the reply. So, suppose I encode an audio buffer (8000 kHz, MONO, float) of 640 PCM frames. In output I have 4 speex frame of 20 byte

Hi All, I am quite new to voice encoding and to Speex library. I have some programming questions that I am not able to answer even after having read the documentation (from the start to the end an from the end to the start) and by reading the sample code (provided in the documentation) and the speexdec.c and speexenc.c programs. I am currently working on an interface between Java code and

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

G''Day I would like to be able to use my VOIP telephone over a saturated ADSL link whilst enjoying optimum audio quality and utilising all of the bandwidth I pay for. It is about this situation that I write. HFSC appears to be the queueing discipline of choice for VOIP. In order for this to work, though, do I have to account for the ATM overhead in the small VOIP packets by defining my

I've targeted Speex to run under eCos with Thumb interworking using arm-elf-gcc 3.3.3 I'd written a small test case to verifiy the operation: #include <speex/speex.h> void speex_test(void) { SpeexBits bits; unsigned int frame_size; void *enc_state; float *speex_frame; cyg_uint64 t1, t2; int i; speex_frame = (float *)0x00030000; /* Address of Speex data

Hello! I was using Speex all the time, and I am now moving to Opus. I had encapsulated the decoder a bit, I had the following cpp file: #include "StdAfx.h" #include "spxcodec.h" #define MAX_FRAME_SIZE 2000 #define MAX_FRAME_BYTES 2000 CSpxCodec::CSpxCodec() : enh(1), rate(8000) { } CSpxCodec::~CSpxCodec() { } void CSpxCodec::Init() { speex_bits_init(&bits);

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