search for: 16khz

...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. One is 8Khz and the other is 16Khz. Both contain identical 4Khz bandwidth audio (the 16Khz file is a properly up-sampled copy of the 8Khz file). When compressing these, the 16Khz input generates about double the opus file size even though the audio content is identical. I found this to be unexpected since I expected VBR mode...

We want to record inbound channels at 16kHz, but send only 8kHz to our peers. I've set our default profile in sip.conf to disallow all but g722, and the peers disallow all but ulaw. We have a proxy in front of Asterisk that is configured to disallow all but G722 also. My test calls show inbound to the proxy is recorded at 16kHz, inbound...

hi on my device i can sample only at 32khz and want to use speex at 16khz so i need to down-convert the input signal by factor of 2. does anyone provide me a reference to some code that does that? are there any trick to do that? i tried to add to subsequent sample but the result was very bad. what are the requrment from a decimation filter for audio? thanks, nir ------...

...o good, though I don't really understand to shift to more system time, and less user time. Also, I suspect this isn't encoding to exactly 28.8kbps, and by using -V I get: I am aiming for maximum dynamic range in my recordings, so 8kHz should be ok, (may even increase dynamic range over 16kHz in the same bitrate?), but I tried it anyway: # time speexenc -w --bitrate 28800 --comp 1 test-16kHz-60sec.wav test-16kHz-60sec.spx Encoding 16000 Hz audio using wideband (sub-band CELP) mode (mono) real 1m46.154s user 1m22.030s sys 0m24.120s Nope, but reducing the bitrate to 20kbps a...

...} Basically you need to write every other byte in thePCM sample stream. Clifton Craig Software Engineer http://codeforfun.wordpress.com clifton.craig at gmail.com On Jun 26, 2010, at 9:50 AM, nir elkayam wrote: > hi > > on my device i can sample only at 32khz and want to use speex at 16khz so i need to down-convert the input signal by factor of 2. > does anyone provide me a reference to some code that does that? are there any trick to do that? > > i tried to add to subsequent sample but the result was very bad. > what are the requrment from a decimation filter for audio?...

Hi all, Can asterisk 1.8.x give me MixMonitor recordings of 16Khz sampling rate? Any help would be appreciated. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120801/ee8ac7a6/attachment.htm>

Hi I have a transcription software which requires media files with wide band frequencies. Is it possible asterisk can record calls with 16khz not 8khz ? Best regards Muhammad Yousuf -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140530/cd3f74c3/attachment.html>

Kirk, Speex was designed for 8kHz, 16kHz, and 32kHz sample rates. If you don't use one of these sample rates, you'll be messing up important assumptions deep within the codec. Why these sample rates? It's telecommunications tradition, rather than PC audio tradition. If you want an efficient and high quality format for v...

Using narrow, wideband, and ultra-wideband encoding on a short 16khz wav gave .spx's of 3,789 ... 2,935 ... and 1,875 bytes. Even after reading the manual, smaller files for the higher frequency encoding seems counter-intuitive. My mp3 at 32 kbps on the original 22khz wav is 3,866 with a quality comparable to speex wideband on the converted 16khz wav, so spe...

Hi, I just released unstable version 1.1.2 that contains more fixed-point work. Though it's still not 100% complete, enough have been done to make it run in real-time on ARM. In order to do that, compile with --enable-fixed-point --enable-arm-asm. All narrowband modes work in real-time with complexity 1 (some work with higher complexity) and some wideband modes also work (up to ~20 kbps) at

The G.729.1 "wideband" codec is starting to show a slight bit of traction. There is a possibility that Asterisk could support G.729.1 - would you use it or buy it if it was available? More importantly, does any equipment with which your systems currently exchange traffic support G.729.1? Currently, the number of devices supporting G.729.1 seems to be fairly limited and it

...from ISR to ISR. With uncompressed audio this works just fine. Now I try to insert OPUS1.3 in the path but cannot make it work. The audio passes through but ends up “down pitched” and heavily distorted. Due to the 32MHz base clock (+ prescalers) available it is not possible to get the desired 16kHz sample rate. I am therefore forced to work with 15625Hz. As I understand it the opus encoder does not mind a strange sample rate as it will re-sample it to 48kHz (Correct?). The decoder, however, will always output 48kHz, 24kHz, 16kHz etc (Correct?) which is incompatible with my 15625Hz rate. (H...

> As you can see Q4 and Q5 are good candidates providing one > frame per block at rates of 12800 and 16800 bits/sec. > SO I'll try the 16kHz modes next, as they should really be an > improvement to quality and clarity. I'll sort out the sources now and > mail them to you. Thanks for your hacked up instcodec source code. I installed the 16.8 kbit/s 16kHz Q5 mode and it worked instantly. It offers a pretty decent and clear so...

...with both testecho and my test program, and for some reason it just doesn't cancel any echoes with the 1.2-rc1. The testecho from beta3 binaries works fine, and also if I replaced mdf.c in 1.2-rc1 with mdf.c from beta3 and use my test program, it will work again. This happens for both 8KHz and 16KHz. Any ideas? I could upload the test samples and results if needed. Cheers Benny -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/speex-dev/attachments/20080809/2cd70322/attachment.htm

I use Speex with dialup modem users. Even if the dialup modem is "56K", you should assume that upstream bandwidth available is only 20-30kbps at the most. I use 16kHz wideband mode and VBR quality 2. Also, I send 80ms (4 frames) per packet, because there is an overhead of approximately 33 bytes per packet due to UDP (8 bytes), IP (20 bytes), and PPP (~5 bytes) headers. If you sent 20ms (1 frame) per packet, this overhead would be 13.2kbps, which is totall...

That's pretty bad. Both DirectSoundCapture and WinMM are capable of recording at 16kHz. I don't know why OpenAL would be incapable of handling it. It's not like it's at all rare or new. I would try 16000 and see if it works. Maybe the docs are wrong? Note that one option to retain high quality is to capture at a higher rate and then downsample using a resampling a...

...e flac-1.1.0 directory into flactest (so that the path flac-1.1.0\include and flac-1.1.0\src exists in flactest). The project should then compile as is. All the code is in flactest.cpp. It's pretty basic. You'll have to create a file test_16k8.wav in the pocket pc's temp directory (16KHz, 8bits). Thanks, for helping, Jehan -------------- next part -------------- A non-text attachment was scrubbed... Name: flactest.zip Type: audio/x-zipped-mod Size: 8384 bytes Desc: not available Url : http://lists.xiph.org/pipermail/flac-dev/attachments/20030523/ea124ff0/flactest.bin

Hello, congratulations to the Ogg Vorbis team - RC2 sounds good. But... RC2 in 128 kbps mode seems to clip off all frequencys beyond 16 KHz. On the tracks I tested Beta 4 gave response even beyond 18 KHz. Some testings on a randomly chosen track: (other tracks gave similar results) Artist: Judas Priest Album: Jugulator Title: Bullet Train Beta4: 127 kbps, ~ 18 KHz (!) RC2: 132 kbps (!), ~ 16

All -- Does anyone have experience doing remote live broadcasts over Icecast? My thought is to use a Dell laptop running Windows (yeah, I know ;-), digitize locally to 16khz, and pump the output to a remote Linux box. Has anyone done something like this before? Thoughts? Issues? Thanks, Roy --- >8 ---- List archives: http://www.xiph.org/archives/ icecast project homepage: http://www.icecast.org/ To unsubscribe from this list, send a message to 'icecast-re...

...nput files that work well with > beta3 and not with rc1? I'll have a look. > > Thanks for the help. The files are on their way now, the upload will take few more minutes to complete. In the mean time let me explain more what I did. The speaker signal is signal-xx, where xx is 8khz and 16khz depending on the file's sampling rate, and mic signal is mic-xx. The result is result-xx-yy, where yy is either rc1 or beta3 depending on Speex version that I used for the test. For the result, I use my test program for the test [1], and during the test I set echo tail length setting to 200ms,...