search for: 4khz

Hi All. I am seeing the following unexpected behavior with OPUS_SET_MAX_BANDWIDTH. I expect that setting this to OPUS_BANDWIDTH_NARROWBAND would give similar results to passing an 8Khz sample rate stream, but OPUS_SET_MAX_BANDWIDTH has almost no effect with any settings. My test data has 4Khz bandwidth. I am testing the opus encoder (latest versions) with the following opus_encoder_ctl options: OPUS_SET_VBR=1, OPUS_SET_VBR_CONSTRAINT=unconstrained, OPUS_SET_COMPLEXITY=10, OPUS_APPLICATION_AUDIO, frame=60ms I compress 2 separate audio streams which only differ in the sample rate....

Hello everyone, I am using the Asterisk REST API in order to establish a call to an endpoint and to send over a remote file (HTTP). The issue is that I am experiencing an audio quality issue. I have tried encoding the file differently, but everytime Asterisk is cutting the audio frequencies above 4Khz. The call is established with G.722 and the audio file is mono 16Khz 16 bit sln16 extension. What can I do to improve the sound quality? Is there any way to not have asterisk cut the audio frequencies? best regards Tiago -------------- next part -------------- An HTML attachment was scrubbed... UR...

...ossible, and a high audio quality demonstration of the difference in pronounciation of a "D" in English, Spanish, Chinese, and German, using as much space as needed to make the difference sound clear. My current thought is to filter the input down to a bandwidth of 7KHz or 4KHz (traditional values for high and low quality speech), decimate the samples so that the sound is sampled at, say, 44/3=14.6KHz or 44/5=8.8KHz, then run it through the standard Vorbis encoder. Vorbis then sees an ordinary 20KHz bandwidth stream that sounds like a tape recording running at 3 to 5 tim...

...e. Basically the problem is that I have an ADSL network that I use to remotely backup some files that have changed across and entire OS every few days. The problem I have found is that banddwidth up is about 50x greater than that coming down (its about 20:1 but add noise on the lower 128KHz of the 4KHz bins used to upstream and in reality it is about 50x). When I sync from the machine at the end of the ADSL connection to anywhere else the file list is always being SENT to by the machine with the slow upstream ADSL connection (swapping the destinations results in a "building file list"...

...than the original (MP3). Of course the encoder sees just WAV (RAW) with possible MP3 decoding artefacts, but anyway: Is there a way to handle this? Maybe applying a low-pass filter before sending to Opus? And why is the 48kHz selected, and why does opus have so few sampling rates (nothing between 24kHz and 48kHz)? The frequency steps for bandwidth are: 4kHz, 4kHz, 8kHz, 24kHz (40kHz in total) With 4,8,12,16 you'd have sample rates 8kHz, 12kHz, 20kHz, 32kHz, 48kHz. Still I'd miss 44.1kHz, though Regards, Ulrich

On Mon, Nov 5, 2018 at 11:01 AM Jan Stary <hans at stare.cz> wrote: Attached I send the spectrogram (vic SoX) of the first 20 seconds > for the wav file and the opus file. Indeed, there is extra noise > for the low frequencies, but somewhere around -100 dB. > > Jan > That might be entirely due to SoX treating it as a 16-bit file, which it is not; -100dB is almost

Thanks! Btw, have you tried using SBR-technology or similar with speech codecs? That might be a good idea I thought.. But I don't know if it produces as good quality with speech codecs as it does for music codecs. Do you know if there is any open source variant of SBR? /Pontus -----Ursprungligt meddelande----- Från: owner-speex-dev@xiph.org [mailto:owner-speex-dev@xiph.org]För Jean-Marc

...er pitched fundamentals which usually give rise to them. I don't know of any voice specific coder that even attempts to capture energy above 10kHz. SBR just isn't relevent. Most wideband speech coding captures only 7kHz to 8kHz bandwidth. The key improvement that gives over the 3kHz to 4kHz most mainstream voice coders capture is to clean up unvoiced sounds. fffff, sssss, and other unvoiced sounds appear almost the same at telephone bandwidth. At 7kHz bandwidth they have enough character to make them more distinguishable. The basic intelligibility improvement you get is usually sm...

In the fir_mem_up filter, the inner loop does a mac on the y0, y1, y2, y3 variables. What is the range of values of those variables? I would like to move the SHR 1 outside the loop after the mac has been done. Will this result in a 32 bit overflow of the y0, y2, y2 and y3 variables? -Fritz On2 Technologies, Inc. http://www.on2.com

Hi all, Does anyone know if Asterisk or any available 3rd party add-on for it support "voice recognition" (not "speech recognition") - task of recognizing people from their voices? Thanks, Alex

...doesn't go any higher than 128KBPS, for example. This problem only occurs with stereo files, not mono. Let's say I set the format to use bitrates up to 45KBPS. Well, the low-pass filter has this problem, the closer the volume gets to 0DB, the lower the cutoff, by 0DB I think it's around 4KHZ cutoff but when the volume gets quiet it sounds like the cutoff is where it should be. The higher I let the bitrate go, the less the lowpass filter acts up, by 80KBPS I can't hear anything wrong with it. On the "dare to compare" section of the vorbis site I heard none of these low-pas...

Hi, I've tried vorbis on a stream that has ~500Hz tops and it worked quite good, but the raw data was converted to a fake 16bit 44Khz WAV Are there any chances I can take the code and tune it to work with nonPCM data that has info from about 5 up to 500Hz? It would be nice to know where to start. Thanks you. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/vorb...

Hi, I have a client that wants a phone system that will play sounds from a sleep machine. I tried using all different formats (GSM, WAV, WV49, MP3 etc.). Over SIP it was OK however with the PSTN it broke up from time to time. I assume this has to do with the fact that the PSTN is limited to 8khz. Is there something I am missing here or is this simply a limitation of the PSTN? Regards,

Hi, I noticed that in some scenarios, Opus 1.2.1 produces better quality than 1.3.1 does. In the use case here, I'm enabling FEC and "transcode" signals from telephony networks (PCMU, 8kHz sampling) to VoIP (48kHz here). In this case, Opus always produced some leakage/ringing above 4kHz but for 1.3.1, these artifacts became worse. The small script below can be used to demonstrate this. Interestingly, the quality is better for the settings in the script when complexity is reduced to 0. I assume the encoder disables FEC at complexity = 0? Is the degradation between 1.2.1 and 1.3.1 a...

Hi, I noticed that in some scenarios, Opus 1.2.1 produces better quality than 1.3.1 does. In the use case here, I'm enabling FEC and "transcode" signals from telephony networks (PCMU, 8kHz sampling) to VoIP (48kHz here). In this case, Opus always produced some leakage/ringing above 4kHz but for 1.3.1, these artifacts became worse. The small script below can be used to demonstrate this. Interestingly, the quality is better for the settings in the script when complexity is reduced to 0. I assume the encoder disables FEC at complexity = 0. Is the degradation a known problem? Thanks...

...which usually give rise to them. > > I don't know of any voice specific coder that even attempts to capture > energy above 10kHz. SBR just isn't relevent. Most wideband speech coding > captures only 7kHz to 8kHz bandwidth. The key improvement that gives > over the 3kHz to 4kHz most mainstream voice coders capture is to clean > up unvoiced sounds. fffff, sssss, and other unvoiced sounds appear > almost the same at telephone bandwidth. At 7kHz bandwidth they have > enough character to make them more distinguishable. The basic > intelligibility improvement y...

...^2). We only care about the lower half of this power (remember we padded with zeroes). Then, let SNR = sum[all i] abs(ref_power[i] / resamp_power[i] - 1.0) IE; SNR = 0 is a perfect signal. Everything else means the signal deviates. There are 3 SNR values posted below. The first value is the 0->4khz range (which for 48khz output means the lower 1/6th of the power spectrum). The second is the 0->8khz range (full original signal), and the last is the full range. The reason I split it is that the filter-based resampler has cutoff filter, so it zeroes out frequencies near the nyquist. So th...

Hi all, I started testing the RC1 encoder at ftp://sjeng.sourceforge.net/pub/sjeng/oggdrop.exe (based on branch_monty_20010708) On the songs I have tested so far (not much :) I did not hear any stereo issues, but there are some very noticeable problems with the produced files. Songs without much high-end will suddenly have one when encoded. (you'd expect it the other way around) It

On Tue, Apr 05, 2005 at 08:26:45AM -0700, Ralph Giles wrote: > AM radio is lower quality (mono) but I don't know > what the digital equivalent would be. Just a minor nit-pick: AM radio can be stereo. However its use is almost nonexistent. See <http://users.hfx.eastlink.ca/~amstereo/amstereo.htm> for more information. > Telephone is nominally 8 kHz mono > (i.e. really bad)

>> Yes, I plan to use it in a VoIP environment if I can get latency reduced to >> an acceptable level :) >> The latency depends directly on the overlap parameter, which also controls >> the quality. Higher quality => higher latency. You could set the overlap to >> 0, but that would give you some nasty artifacts. >> You can also resample with smaller block