similar to: Speex quality estimation in lossless media

OK. Different mailing lists are set up differently. This list is unusual because your answers only go to the person who replied to you. So if you want the other people on the listserv to see your answer, you should make sure that Speex-dev at xiph.org<mailto:Speex-dev at xiph.org> is added to the TO: field of your outgoing message. Hopefully someone else will also attempt to answer your

Ben Greear wrote: > Francois Menard (Mailing List Account) wrote: > >> >> did you try speex in wideband mode ... what bitrate? >> >> f. > > > 15kbps mode. It does significantly better with a different > speaker (the male does better than the female), as well. Most CELP based codecs do a better job with lower pitched voices, so male voices tend to sound

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

Francois Menard (Mailing List Account) wrote: > > did you try speex in wideband mode ... what bitrate? > > f. 15kbps mode. It does significantly better with a different speaker (the male does better than the female), as well. I'm considering purchasing a commercial PESQ license and wrap the PESQ software in a server application. I could then allow other folks to run PESQ

Hello, I just signed up to this mailing-list (note: my first mailing list at all), because I'm having some problems related to speex. Let me just introduce you to what I'm doing. I am writing a short (really short) paper about VoIP techniques, especially audio codecs for speech. I pointed out basic technologies behind audio codecs; vector quantization, lpc, long-term prediction and some

Jean-Marc Valin wrote: >> What commercial VOIP test products can I turn to in order to objectively >> evaluate your CODEC; in order to analyze audio streams for MOS scores and >> performance parameters? >> > > The MOS evaluation is subjective, not objective and actually involves > getting (many) real people to listen to the audio, not just buying a >

Hi Jim, First of all - thanks, turning the highpass filter off was what I needed, and the waveforms match now. But, when i did the PESQ tests again I found an interesting result : version 1.0.5 still got a slightly better average score, but the standard deviation on version 1.2 beta1 was much smaller. The cause for that is this - on some samples versions 1.0.5 and 1.2beta2 produced a single

Hello! Some time back, I added the Speex protocol to my version of VOCAL (www.vovida.org, VOIP tool). Recently, I also added PESQ (automated voice quality testing algorithm) to my tool and have been running some tests on a clean network. The source file is a woman reading some phrases meant to test various aspects of codecs... Speex has a respectable result of 3.67 Some other codecs I've

Hi, I'm using the ITU-T P.834 methodology to derive the Ie parameter for Speex. According to the E-model, Ie indicates the degradation in quality of a codec at 0% packet loss. It is a non-negative value: the higher, the worst. Details in ITU-T G.107. Because Speex is multi-rate, we are deriving one Ie parameter for each Speex rate. In our preliminary experiments, we've got a value of

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

Hello, What is meant by crude delay in PESQ quality testing? And what is its tolerant limit for good quality. rgds, Sharan

Hello developers, I have a question about supported bitrates for SPEEX. Regardless of whether I specify a 10kbps or a 8kbps bitrate, the encoded .spx file is the same size and have the same PESQ value. This makes me wonder if SPEEX supports arbitrary bitrates, or only a certain set. A few more details about my configuration: I am using SPEEX encoder version 1.1.12 I am encoding a narrowband

Hi, I have been using speex version 1.0.5 on a text-to-speech program. Recently I upgraded to version 1.2beta1 and noticed that the waveform the I got after encoding and decoding on the new versions (beta1,beta2) is much more different than the original than on version 1.0.5. I also ran a PESQ comparison test on 700 voice samples and got better results in the older version (I used quality 9, and

This is regarding the speexdec.c of the latest beta2 version. After decoding a frame, for packet_no =1, a new_frame_size and frame_offset are calculated and the number of decoded bytes writting onto a file depends on them. Why are they calculated? What happens if i directly write all the 320 shorts I get after decodong onto a file? --sharan -----Original Message----- From: Jean-Marc Valin

Overview Asterisk-powered dialer software Web Interface UNIX/Linux Cron-based Schedule Logic Open-Source Code Graphing Monitoring Stats MySQL Database for Call Records Current Features Dial by SIP or PSTN - Asterisk base capable of dialing via any medium Blast-Dialing - send multiple calls to 1 trunk for specified duration - No QoS/MOS scoring performed, designed for load testing

Hi folks sincere apologies if this is not meant to be posted here. I am the CEO at http://directi.com. One of our products http://talk.to is in the messaging space and we are planning to add voice support to our application and are currently investigating codecs (Opus, ilbc, isac, G.729, AMR etc). There are several variables and obviously Opus would be our preferred choice given its quality, VBR,

Hi folks sincere apologies if this is not meant to be posted here. I am the CEO at http://directi.com. One of our products http://talk.to is in the messaging space and we are planning to add voice support to our application and are currently investigating codecs (Opus, ilbc, isac, G.729, AMR etc). There are several variables and obviously Opus would be our preferred choice given its quality, VBR,

Wang, Chris wrote: > Benjamin, > > Thanks for the prompt response. Are there other recommended methods to > verify encoder implementations? There are few if any requirements placed on the encoder by the actual RFC, however there are a number of things you can do which provide various amounts of assurance your encoder is working as expected. The first test I would run is to write

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

Hi Aviv, Does the audio sound bad? You can try turning off the highpass filter (which was not in 1.0.5). This code is from ti/testenc-TI-C5x.c in the source tree: /* Turn this off if you want to measure SNR (on by default) */ tmp=0; speex_encoder_ctl(st, SPEEX_SET_HIGHPASS, &tmp); speex_decoder_ctl(dec, SPEEX_SET_HIGHPASS, &tmp); - Jim ----- Original Message ----- From: