similar to: Opus encoding rate for very quiet noisefloor

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

Hi Opus list, I'm noticing a discontinuity in the quality between use of 10ms and 20ms framesize for mode=AUDIO at a bitrate of about 28000. Quality drops audibly for voice signals when encoded at 20ms framesize, versus quality at 10ms. This effect is mode=AUDIO only. Using mode==VOIP shows no sig. difference between 10 and 20ms framing at this bitrate. Pesq totally

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

subject line says it all. I am trying to use the Google Storage gem (gstore), and if i use ruby 1.8.7, no problem. If i try to use ruby 1.9.3, i get this error: Timeout::Error: Timeout::Error from /usr/local/lib/ruby/1.9.1/net/protocol.rb:146:in `rescue in rbuf_fill'' from /usr/local/lib/ruby/1.9.1/net/protocol.rb:140:in `rbuf_fill'' from

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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

Hi, I am using speex 1.2beta2 on a 24bit DSP that has a severe program and data space limitations. I am only interested in the speex decoder for 11kbps narrowband implementation. I am using the following parameters and structures 160, /*frameSize*/ 40, /*subframeSize*/ 10, /*lpcSize*/ 17, /*pitchStart*/ 144, /*pitchEnd*/ /* 11 kbps medium bit-rate

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, 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 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);

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

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

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, Can anyone let me know if this is a bug? Thank you, Chandrakala ----- Original Message ----- From: "Logan Stromberg" <loganstromberg at gmail.com> To: "Chandrakala Madhira" <chandrakala.madhira at soctronics.com> Cc: opus at xiph.org Sent: Wednesday, November 22, 2017 12:12:39 PM Subject: Re: [opus] Reg an issue with smoothing factor in VAD

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

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 have a simple question about speex frame size. i work with jspeex - but i think it's speex generic, the question i have. can i use any frame size i want? or does the size have be in a certain ratio to other numbers? and what does the frame size in the decoder mean? encoded or decoded packet frame size? how much data can i hand to the encoder to encode? if i hand i.e. 400 bytes to

Hi Jean-Marc, Sorry for late reply, thanks for interest. It's quality good for 10ms/audio, poorer for 20ms/audio. Quality equivalent for 10,20ms for mode=voip. PESQ was the tool that alerted me to something of interest, but I don't trust PESQ to almost any degree! It's good for hearing relative differences, of course, but not absolutes. Bitrate here was 28kbps, but I hear

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, 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