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

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, 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

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

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

An HTML attachment was scrubbed... URL: <http://lists.xiph.org/pipermail/opus/attachments/20190514/f9c47ab4/attachment-0001.html> -------------- next part -------------- A non-text attachment was scrubbed... Name: 201602111742151_N3WZA6X7.png Type: image/png Size: 33527 bytes Desc: not available URL:

Just for fun, I tried to reproduce such an overflow. I turned on all debug macros, assertions, and checked arithmetic and then encoded 2 hours of mixed speech/audio with these parameters: Sample rate = 48000 Channels = 1 Application = OPUS_APPLICATION_AUDIO Bitrate = 24 KB/s Force Mode = MODE_SILK_ONLY Signal Type = OPUS_SIGNAL_AUTO Complexity = 10 Frame size = 480 samples (10ms) No errors came

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

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

On Sat, 2015-10-24 at 22:16 -0700, Thomas Daede wrote: > Everything above 96kbps on that table is speculative, as the highest > multi-participant listening testing done was at 96kbps. Here's the > results from that test, if you're curious: > > http://listening-test.coresv.net/results.htm > > As you can see, at that rate Opus ranged from slightly perceptible to >

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 Chandrakala, Logan, Can you confirm that the attached patch fixes the overflow problem? Koen, can you confirm the fix makes sense? Cheers, Jean-Marc On 11/27/2017 12:10 PM, Logan Stromberg wrote: > Sorry, long holiday weekend in America. > I can say with pretty high certainty that there is an overflow occurring > and it is flipping smooth_coef_Q16 to be negative when it probably

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

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 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