similar to: Listening tests -- please participate

Ron wrote: > I'm curious about that because you note that the Opus bitstream > wasn't designed to be inherently robust against a bit error at > any point in the packet (and some places will obviously effect > it worse than others) - however it's not quite true that it > 'cannot tolerate them'. It's not quite true to say that Opus was not designed to be

Hi Jean Marc, I was very impressed by the comparative results. Could you give a bit more information about what really are 7kHz and 3.5kHz in the 48kHz table? I am looking forward that this will bring a bigger separation between the basic tools and speex in a sense that the basic tools could be used with more than one codec. I even understand that there is interest in the tools alone (Echo

As far as I can see, transcoding could be usefull for people who do not primarly care about quality but about filesizes. One could assume that such a user would have a collection of mp3's at 128kbps or higher bitrates, and uses an encoder like BladeEnc or Xing. He wants to take uses of ogg's supposed quality and transcode his 128-or-higher files into 96 or 112kbps oggs to save diskspace.

Hello all, This is slightly off-topic, but should be of interest to some people on this list. I just released version 0.3.2 of the CELT ultra low-delay audio codec (http://www.celt-codec.org/). CELT is designed to encode high quality speech and music with less than 10 ms delay and at rates starting from around 32 kbit/s. This version is "special" in that it is the basis for some

You know it's good; I know it's good. I'm talking about Vorbis at 128 as it currently is in CVS. Please participate in a group listening test of various formats to show how Vorbis 128 has improved since RC2. I have prepared three sample music clips comparing Liquid AAC, MPC, pre-RC3 Vorbis, Lame, Xing, and WMA8, similar to the first test. Except I believe that this time Vorbis

I'm trying to convert my current headphones and microphone chat application to support loudspeakers and microphone, and so I thought I'd give speex_echo_cancel() a try. Since my users quite frequently have other sound-producing applications running on their computer (such as winamp), I sample 'wave' recording device of the soundcard in addition to the microphone. I then call

Hi, A couple things you may want to check: - set sampling rate to 8 kHz (at least for now) - make sure the far end signal in the playback signal is always a bit in advance (never late) compared to the mic signal. - Set the tail length to something around 100 ms. Also, if you're using two different soundcards (as I understand) for the playback and the capture, you're *never* going to get

In our experience, the total cost of softphones(money, reduced sound quality and lower reliability) in a large call center environment is actually greater over time than the cost of a channelbank and cheap analog headphones. We've tried 2 softphones, 2 kinds of SIP VOIP hardphones, 2 kinds of SIP analog adapters and we've tried channelbanks over the last 3 years. Right now we are half done

Dear friend, We are conducting a study on the motivation of the knowledge sharing on the CentOS community. The contributors? experience to Linux is very important to the design and management of this knowledge platform. Would you please post the following on-line questionnaire message to the CentOS platform or forward the message to the members? After the survey is done, we will randomly select

Hello everyone! Sorry, if the whole task is silly, I'm new to this. I have my newly installed asterisk (1.6.0-beta9) and my AVM Fritz a1 card. I have a simple German isdn line and I have a microphone, headphones and a running JACKd (JACK Aduio Connection Kit). The question: Can I (mis)use my asterisk CLI interface to make and recieve calls coming in/going out via the ISDN-card,

Do you know of anyone, especially in the free software world, who has successfully tackled the problem of synchronizing real-time input and output audio streams of different devices using Windows? I need to do a good job of this so that Acoustic Echo Cancellation can work in my Video Conferencing application. I need to be able to capture audio from a USB webcam and play it out over the user's

This afternoon, I bought a pair of ear plugs (the large flat kind that hook onto my ears). Plugged them in to test and they work perfectly. I'm listening to streamaudio.com as I type this. However, when I tried to watch a youtube.com video (my wife exercise dancing) just silence. She had the same video on to test and the sound was fine on her box. Then, I tried system-config-soundcard and

Thank you. Very helpful. > On Jan 27, 2017, at 12:40 PM, Jean-Marc Valin <jmvalin at jmvalin.ca> wrote: > > On 27/01/17 12:29 PM, Jon Lederman wrote: >> Thank you. Yes, we do need both channels independent. So, if we >> encode each channel separately, we will be sacrificing the >> compression ratio we would achieve with stereo encoding, correct? > > Not

Hi All, I was wondering whether there have been some listening experiments done to test how well spatial information is preserved in the celt signal, e.g. comparison of sound localization performance for the original uncompressed sound and the celt sound (most probably for different bit rates). Best, Pablo -- Pablo F. Hoffmann PostDoc Acoustics Dept. of Electronic Systems Aalborg

On 27/01/17 12:29 PM, Jon Lederman wrote: > Thank you. Yes, we do need both channels independent. So, if we > encode each channel separately, we will be sacrificing the > compression ratio we would achieve with stereo encoding, correct? Not necessarily. Stereo makes two assumptions: 1) It assumes the two channels are somehow correlated 2) It assumes the two channels are meant to be

Hi! What are the assumed listening conditions in vorbis ? For example there are big differences in perceived sound when listening with headphones compared to listening with speakers. ( "super stereo effect" , different frequency response due to HRTF etc ... ). Is this stuff worth considering in a vorbis class encoder ? David Balazic --- >8 ---- List archives:

> As the thing is to encode for human ears (AFAIK), I'd say that 4kHz is already "quite high", > and I wonder who can actually hear pure 20kHz sine. If you read the beginning of RFC 6716, you learn that Opus never encodes any frequencies that are higher than 20 kHz. So at some medium or high bitrates, anything above 20 kHz is filtered out, not because of the bitrate but

Hi Christian, Thanks for publishing this. Do you happen to have some numbers on what the expected bit error rates are for a "typical AMR-WB usage scenario"? I'm curious about that because you note that the Opus bitstream wasn't designed to be inherently robust against a bit error at any point in the packet (and some places will obviously effect it worse than others) - however

I've tried almost any softphone available on the market with many different PC, soundcard, headphones combinations. None of them prooved production reliable in a call center environment. I've also tested many IP Phones and Gateways. Even the cheapest one supplies much better quality. Is this a fact or am I missing a point. I would certanly prefer a softphone because of cost and

Enrico Minack wrote: >>I tried, and I noticed that it works fine for mic, line1... pressing >>space moves the "R" to the selected line... but it doesn't work for >>pcm or master: so it seems to confirm that the soundcard doesn't >>support that... > > so it seems its a half-duplex sound card. what about using a audio cable > between headphones and