search for: frame_sizes

Hi, I'm using the echo cancellation api and I would like to clarify the 'frame_size' parameter used in speex_echo_state_init(frame_size,filter_length). In the 'docs' it says: "...where frame_size is the amount of data (in samples)you want to process at once..." So... here are my questions: if I use stereo input/output do I have to put the samples doubled ? For

Hi Jean, Thank you for your reply. Ok... I'm gonna use 'samples per channel' everywhere I see 'samples'... but what about the 'speex_echo_playback' function ? it does the following loop: ... for (i=0;i<st->frame_size;i++) st->play_buf[st->play_buf_pos+i] = play[i]; ... So... if frame size is 'samples per channel' it will copy only half the

-------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 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>

I have been working on a project importing Speex to a embedded platform for a while. When I tried to shut down the macro TWO_PATH in Speex AEC to save space, an error occurred. Dbf is undefined. The 1.2rc1 code is like this Dbf = 0; See = 0; #ifdef TWO_PATH /* Difference in response, this is used to estimate the variance of our residual power estimate */ for (chan = 0; chan < C;

Finally got it to compile and attach the MSVC debugger. It fails at the following: if ((!st->variable_duration && 400*frame_size != st->Fs && 200*frame_size != st->Fs && 100*frame_size != st->Fs && 50*frame_size != st->Fs && 25*frame_size != st->Fs && 50*frame_size != 3*st->Fs) || (400*frame_size <

"frame_size = PCM audio in signed 16-bit format (native endian). There must be exactly frame_size samples per channel." Sorry for such a simple request, but could you please clarify the definition of frame_size when using stereo audio channels? When you say "frame_size samples" does this mean that, if my frame_size is defined as 128, there are 64 left samples and 64 right

I have problem with coding and decoding. I record voice from mic coding and write to file. And when I read from file and put it on sound card it's terrible sound. Have sameone can help me? #include <speex.h> #include <stdio.h> #include <fcntl.h> #include <sys/ioctl.h> #include <sys/soundcard.h> /*The frame size in hardcoded for this sample code but it

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

Hello everyone I've a following problem with the speex codec. I'm developing application that is supposed to read sounds from the microphone with DirectSoundCapture. Unfortunately when I want to compress and decompress the sound buffer from DirectSound I only have jitter. Maybe you have some code samples how to compress standard DS buffers: LPDIRECTSOUND8 lpDS = NULL;

Thank you for the quick response Thorvald, but I think that's not the problem here :[ I know how to capture the buffer and how to play it in the output buffer of the DS. The problem is (probably) with same kind of short/floats/bytes error in conversion/copying that the coder doesn't get. I can have my buffer locked during the compression, it not the problem at the moment. What I really

Attached is the amended second patch. It now extends the multistream API as well to 80/100/120 ms and incorporates changes based on Mark's comments. Thanks, Felicia On Mon, Jun 13, 2016 at 4:21 PM Felicia Lim <flim at google.com> wrote: > Hi Mark, Jean-Marc, > > Thanks for your comments. > > On Sun, Jun 12, 2016 at 6:34 AM Mark Harris <mark.hsj at gmail.com>

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 Ulrich, thanks for the suggestion. My concern is that one of the valid inputs is "2.5", which would require conversion to an int, e.g. x10, but doing something like this would start to affect the code readability. On Mon, Jun 27, 2016 at 3:02 PM Ulrich Windl < Ulrich.Windl at rz.uni-regensburg.de> wrote: > Hi! > > A note on style: Looking at this chunk of the patch

In stereo mode, you need to use the init_mc() call and consider the number of samples *per channel*. Also, more than 20 ms frames are a bad idea. Jean-Marc danflu at uninet.com.br a ?crit : > Hi, > > I'm using the echo cancellation api and I would like to > clarify the 'frame_size' parameter used in > speex_echo_state_init(frame_size,filter_length). > > >

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

hello everybody in this great mailing list , i have some difficulties to follow my code . i solved some problems thanks to Carine Liang , but i still have one problem and i think it is fatal one. when i encode the voice data in a wav file it is decoded without any errors it gives me 84 bytes wav file size for 139 kbytes wav audio data .Naturally i wanted to return my file back to its normal

Hi Steve, thanks for the advice. I placed the appropriate functions outside the while statement but I still get this error at the while loop: First-chance exception at 0x004010fa in SPXDEC.exe: 0xC0000005: Access violation reading location 0x0000000c. Unhandled exception at 0x004010fa in SPXDEC.exe: 0xC0000005: Access violation reading location 0x0000000c. It looks like the same error. I've

hey guys, I just compiled an application similar to sampledec.c (for speex 1.1.10) and it was fine but when I executed it, the app exited without doing anything. I'm using MS VC 6.0 and this was all I got - First-chance exception in sampledec.exe : 0xC0000005: Access Violation. Has anyone encountered this / does anyone know how to deal with it? by the way, sampleenc executed perfectly... When

Hi, I would like to have some more information about the parameter of speex_echo_state_init. I have a clear idea of the importance of the filter_length in the echo cancellation, what it is more obscure for me is the role of frame_size and how its value affect the algorithm. Let use, for example, frame_size = 400, would I obtain the same result of using frame_size = 200 but with half method call?