similar to: CELT 0.11.1 minimum Frame Size

Hi all, In my setup, I'm using a framesize of 320 samples at a sample rate of 32000 Hz (10 ms frames). This gives me a mode where nbEBands is 21 and eBands[21] is 100 (the standard 2.5 ms short block ebands). This also gives me LM = 2, so what happens is eBands[nbEBands]<<LM is 400, whereas my framesize is only 320. This causes a stack overrun at line 1978 in celt.c in both calls

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

Hi everyone, I'd just like to announce CELT 0.11.1 (sorry about failing to announce 0.11). The bit-stream is now is "soft-freeze", which means we won't change it unless we find bugs -- which we did between 0.11 and 0.11.1. The API should also be close to the final API. Compared to 0.10.x there are also small quality improvements, mostly at high bit-rate. I'd also like

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

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);

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

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 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

I have build celt with FIXED_POINT option (latest 0.6.1 as well as from the git repo) on windows. However I am not getting a valid output (all samples are saturated) when I try to decode with this version. The input file was encoded with the same fixed point version. Does fixed point version work at all? Following are the command line settings; Samplerate; 48000 Channels: 2 Framesize: 256

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

Hi Kevin, Are you saying that the quality is good at 20 ms and bad at 10 ms, or the reverse? Also, is this speech or music? What tool, what options? In general, it helps a lot if you post the sample (input and output). Cheers, Jean-Marc On 06/03/2016 12:48 PM, Kevin Connor wrote: > Hi Opus list, > > I'm noticing a discontinuity in the quality between use of 10ms and > 20ms

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

Hi I have just written a small c programm which reads in speex files and outputs them as asci text numbers. This is necessary as for writing a hw testbench for decoding tests, since thy only read ascii files :-(.I can't read the bitrate properly from the header file. It's allways -1. Please see example below and attached c programm. I wan't to check for a fixed bitrate (8000).

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

Hi everyone, I've recently been making various changes to the way the modes work and the supported frame size. On new feature that may be of interest to some is that CELT should soon support changing the frame size dynamically within a stream. By that I mean varying the amount of audio (in time) transmitted at once, not the compressed size -- which has always been variable. That would

Hello, I am interested in using the CELT codec for an embedded project around the DSP C55x family from TI. Is this someone has already done? Which is necessary to do that, which compiler, is there a specific configuration ? Is it necessary to have an Operating system to run the CELT codec ? Is it possible to instantiate several times the CELT codec ? Thank you in

Hi all. I'm trying to decode speex using version 1.1.10's libspeex with fixed_point enabled. copied the sample in the manual (1.1.11) with minor revisions. While running the program, encountered this warning: "Packet is larger than allocated buffer : 38" when calling speex_bits_read_from (&bits, cbits, nBytes) then my program terminated unexpectedly with errors (Unhandled

Hi, I have found differences in quality between opus_demo and opusenc/opusdec. I used for both applications the same raw pcm file,16 bit,48khz,litle endian. i use libopus 1.1-beta and opus-tools-0.1.7. The command for opus_demo is: opus_demo audio 48000 1 64000 -cvbr -framesize 10 in.pcm out.pcm For opusenc/dec: opusenc --raw --raw-chan 1 bitrate 64 -cvbr --framesize 10 in.pcm in.opus

Hello everyone, I was curious if anyone has performed any listening tests or done work with solo instrument performances with the CELT codec? I experience quite a bit of noise coming up with a heavily tonal stimulus, the worst offender being a trumpet. I understand some of this behavior is to be expected, this is a perceptual codec afterall, but was curious if anyone was able to find an elegant

Hi, Can someone please tell me how should I go about changing the frame size which is hardcoded to 160 for NB and WB and 320 for UWB. For NB speech(8KHz) the framesize of 160 is 20ms frame but for WB and UWB its 10ms. What are the parameters being affected by simply changing the framesize and sub-frame size in "modes.c" How to change the buffer size and how its affected. can we have a