search for: framesizes

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

I have been using the Speex NB mode with an 11025 kHz signal, without adjusting the framesize, and it's been sounding just fine to me. However I was wondering what the best way to make 11025kHz would be. I also have an audio driver that uses 2kB buffers, and the NB framesize of 160 doesn't divide evenly into that so I end up having to shovel tails of buffers around which isn't too

>jesus: > Now I see that it happens when I specify a framesize in > speex_echo_state_init greater than the one used in > speex_preprocess_state_init. > > Next combination works: > codec framesize = 160 > preprocessor framesize = 160 > eco framesize = 160 > > > Next combination DOESNT work (produces heap corruption): > codec framesize = 160 >

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);

Oops... You've just found a bug. Seems like you're the first one to use that call. Anyway, it's now fixed in CVS (both trunk and 1.0.x branch). Thanks for the bug report. Jean-Marc Le mar 03/06/2003 à 01:16, Christian Buchner a écrit : > Hi there. > > I am having a little trouble understanding the frame sizes chosen > by the codec. > > testenc_uwb.c from

Hi, I've been trying to encode a live audio input from the microphone on iOS device using opus. Uncompressed audio recording works fine with http://theamazingaudioengine.com/ Then, when I tried to do encoding, I'm stuck at figuring out why the buffer is empty: static int ec_write_byte(ec_enc *_this,unsigned _value){ if(_this->offs+_this->end_offs>=_this->storage)return

...ng > at this bitrate. Pesq totally overreacts, as it is wont to do :) > However, I do hear a slight drop. Admittedly, 28kbps is a low bitrate > to be running in mode=AUDIO. Is this effect known? Is there a > difference in processing with audio mode between 10ms and (other > framesizes)? I reckon it will go away if I throw some more bitrate > at it, but wanted to understand it a bit better. > > Thanks very much, KevinC > _______________________________________________ opus mailing list > opus at xiph.org http://lists.xiph.org/mailman/listinfo/opus >

Hello, I know my question has been asked before because I spent the last week searching the web for how to use Speex in combination with WaveIn/WaveOut and I ran into a few posts, but none of them answer the question. There is still a lot of confusion how to use WaveIn/WaveOut and Speex by junior developers such as myself. Even after examining code for SpeexDec and SpeexEnc, I cannot get clear

I found a problem with trying to use the speex plugin for xmms on my PowerBook G3 'Pismo' PPC machine, running kernel 2.4.19. It turns out that the kernel sound driver ignores the endian of the data format flag and instead expects the data to be native endian. This is the same problem that was affecting speexdec. I have now fixed this problem, and another compilation issue that I found. I

I didn't expect this, is it normal? When you use inbandfec, and have packet losses, you end up with more audio samples than you started with. With short frame (2.5), the FEC isn't supposed to do anything but it shouldn't do that should it? and 20mS frame, should do 'something' but? add audio? I'm using opus_demo for this. Here's the steps (I have my test audio in

Hi, when I use preprocessor with AEC, VC++ alerts me about heap corruption. I have protected speex_echo_playback, speex_echo_capture and speex_preprocess_run with mutexes, to avoid echo_state being used at the same time, but it still happens. Any help about this ussue? Thank you.

Hi Jean-Marc, Hi Jim, I have also seen some problems with the 1.1.8 release on the C55x. So far I have boiled down the issues to the following: 1) We need our own "fixed_xx.h" header file. I don't know why, and haven't had time to investigate, but there is a definite improvement when I use the attached fixed_c55x.h file which has turned all the maths into inline functions.

Hi Jean-Marc, For the residue in the echo_cancel function I pass in a pointer to an array of floats. First I used an array the size of the framesize, but I discovered this causes crashes because the echo_cancel funtion writes framesize+1 floats to the pointer. I find this plus one strange, since almost all things are exactly the framesize. Is this an error in the echo_cancel function or simply

is there any particular reason why there is no option to configure the codec framesizes in iax2 ? It would come rathrer handy to decide if you want less bandwidth or more robustness on the payload side ... -- Michael Bielicki Managing Director TAAN Consultants Ltd http://www.global-gateway.net/ -------------------------------------------------------------------------- This corresp...

Thank you for such a quick response. The only reason I started with Char buffers is because WaveIn and WaveOut on Windows XP accept/emit WAVEHDR structures, which store audio data in LPSTR, which is Char*. typedef struct { LPSTR lpData; DWORD dwBufferLength; ... } WAVEHDR; When I was going from Char to float and back looked very wrong to me as well, but I was just not

Hello all.. Im new to speex, and im trying to use it in my application to transmit speech over multicast. I've done some test, but i have some questions. First, I'm reading audio with a buffer of 512 bytes, but the speex frames are 160 (using SPEEX_GET_FRAME_SIZE), so i have to manually divide the 512 bytes (256 16bit samples) in blocks of 160 float samples. Using 1 frame, i dispatch

> Oops... You've just found a bug. Seems like you're the first one to use > that call. Anyway, it's now fixed in CVS (both trunk and 1.0.x branch). > Thanks for the bug report. Hmm, the call to speex_encode_stereo() takes the frame size as an an argument - and this call generates bits ;) So possibly my bitrate table shows wrong values in the stereo cases. Hmm... Where would

int frameSize; speex_encoder_ctl(enc_state, SPEEX_GET_FRAME_SIZE, &frameSize); Is frameSize in bytes? If not, what unit is it in? I need to know so I know how big to make each element in my circular buffer. Also, do I need to call speex_encoder_ctl( enc_state, SPEEX_SET_SAMPLING_RATE, &sampleRate ); Depending on sample rate I record my audio in? If not, is there any benefit to

--- Miroslav Lichvar <lichvarm@phoenix.inf.upol.cz> wrote: > I have few notes: > > It seems there is changed API in CVS again. So, what about adding > function like > unsigned FLAC__format_frame_size(const FLAC__Frame *frame) > which returns size of the frame in bytes. This can be useful, for > example, in xmms plugin to display current bitrate like vorbis plugin >

--- Miroslav Lichvar <lichvarm@phoenix.inf.upol.cz> wrote: > On Fri, Nov 08, 2002 at 12:39:52PM -0800, Josh Coalson wrote: > > --- Miroslav Lichvar <lichvarm@phoenix.inf.upol.cz> wrote: > > > I have few notes: > > > > > > It seems there is changed API in CVS again. So, what about adding > > > function like > > > unsigned