search for: 8khz

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 in Asterisk is only...

Hi,I have been using wav files with sample rate of 8khz and 8 bits and I find the sound quality really poor.What is the best sound quality I can achieve on Asterisk?Responses would be appreciated.Rgds,Akpome _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE! http...

Hello Mailing List, I am a Speex supporter and user that would really like to know how much memory Speex uses to decode a 8kHz, 16kHz and 32kHz (primarily the 8kHz) and is it possible to use a 1kBytes of RAM to decode a 8kHz stream? (I was thinking of the possibility of using a ATmega168 to decode Speex) //P?r, Sweden

...especially interested > in feedback about how it behaves when decoding files that were encoded > with a floating point version or vice versa. Encoding on an XScale-PXA255 @ 400MHz: First I tried narrowband, at 24kbps, with verbose output: # time speexenc -n --bitrate 24000 --comp 1 -V test-8kHz-60sec.wav test-8kHz-60sec.spx Encoding 8000 Hz audio using narrowband mode (mono) Bitrate is use: 18200 bps real 1m8.042s user 0m48.240s sys 0m11.220s I guessed the verbose output was slowing it down, so: # time speexenc -n --bitrate 24000 --comp 1 test-8kHz-60sec.wav test-8kHz-60sec....

Hello list, I am about to have to dump Asterisk in favor of some other VOIP/PBX solution; the reason? I have 304 voice prompts recorded as 22Khz wav format files that sound like crumpling paper whenever I convert them to the 8Khz wav/gsm format required by Asterisk. I was considering trying the G.729 codec, but reading through the specs, I see that the 8Khz conversion is going to dump me into the same pile of dung. Any body have any suggestions? Thanks Danny Nicholas -------------- next part -------------- An HTM...

Hello, I'm trying to find simple MP3 player in flash, to integrate it with call recordings. My requirements would be: * simple UI * buffering (would be nice) * slider * volume control * support of 8kHz stereo mp3 * javascript access to seek/position * free for any use (GPL, MPL, MIT, BSD) So far I've found that JWplayer[1] does great with my recordings. However it's not small in size, as there's video player and playlists - none of which i need. Also it should be paid, even if i use...

Hi, is there any way to tell the decoder the output sampling Fz we want ? opus_decoder_create = Sampling rate of input signal (Hz) Considering this example (VoIP-out from WebRTC/RTP) MICROPHONE(44.1/48kHz) >> [encoder created at 48kHz but with internalSampleRate set to 8kHz]>> INTERNET >> [decoder(created with 48kHz)] >> 48kHz(?) >> G.711(8kHz) This leaves us with the only option to re-sample even if the internal sample rate was set to 8kHz. This may not seem like a b...

Is there support for encoding/compressing 8KHz sampled speech? Am I heading in the right direction with Vorbis? I have heard results with 44KHz audio and am quite impressed. As always, any help appreciated. Regards, Paul -- Paul McHale Work: 937-320-5495 Double E Solutions Mobile: 937-371-2828 1435 Edenwood Dr...

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

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 forma...

Hi, another question on the same topic Speex resampler at 44.1kHz seems to be very CPU intensive on Android (even more than the Opus encoder) While Speex at 48kHz is just fine. I wonder any alternate solutions or ideas ? Improve it, look for alternate solution ... I am guessing the NEON optimization are still used for both, etc. On Thu, Apr 2, 2015 at 4:46 PM, Jean-Marc Valin <jmvalin at jmvalin.ca> wrote: > The encoder and decoder can handle,...

Please consider using 16-bit 16kHz (wideband) instead. It's a huge increase in audio quality and the bitrate is still very low, especially if you take advantage of Speex features such as VBR. 8kHz seems totally inappropriate to me for desktop streaming audio, let alone 8-bit samples. Or perhaps your recording equipment is an original Sound Blaster from 1989? (Even that could record at 12kHz.) People often tell me how amazed they are with the audio quality of my VoIP software based on S...

I am wont to decode a speex 11kbps 8kHz 16bit to a raw data 8kHz 16bit LSB on a c5509. Trying to understand the "testenc-TI-C5x.c" exsample, but it looks to me wary complicated. Is there more documentation for the exsample or a decoder exsample available? Can somebody help? Peter -------------- next part -----------...

...specs: Size: 13603893 bytes Header found at: 0 bytes Length: 3733 seconds MPEG 2.5 layer 3 29kbit (VBR), 142925 frames 11025Hz Mono CRCs: No This 13mb file has very acceptable voice quality and plays for over 100 minutes I think. The vbr bounces between 8 and rarely 32 kbit. I believe that even 8khz Mono would be satisfactory for such voice but my understanding is that ogg doesn't go that low. Now that speex has joined xiph, will ogg be able to handle the low end for speech? Being able to get 80-100 hours of speech onto a cd would be a boon to economically sharing. Anything like that po...

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,...

Hi I am using Jspeex for my project which requires compression of audio in realtime..so far i managed to capture sound using java's sound.The capturing format i use is 8 bit 8khz ,stereo pcm.The captured sound is buffered and fed to encoder.Its fed to decoder,whose output is again buffered for sometime.But when i try to play it back i could hear sound only in right channel.Left channel is fully noise.Stereo mode was used in encoder and decoder.I am not using any format conv...

...ith amplitude 0x6000 and frequency about 1150 Hz) E. encode the PCM data and decode it immediately, count the CPU usages. F. repeat until reach the duration time (1000ms or 10000ms) G. The summary of STM32F407 Test Result as below: Mode Sample Chan Freq. Duration Encode + Decode = Total FLOAT 48kHz 2 1150 1000ms 2735ms + 3367ms = 6102ms FIXED 48kHz 2 1150 1000ms 2112ms + 1543ms = 3698ms FIXED 48kHz 1 1150 1000ms 1312ms + 911ms = 2249ms FIXED 24kHz 1 1150 1000ms 1067ms + 783ms = 1872ms FIXED 16kHz 1 1150 1000ms 922ms + 711ms = 1651ms FIXED 12kHz 1...

I attach two wav files of a few seconds duration that fail to play on Windows Media Player 11 using DirectShow Filter bundled in oggcodecs_0.71.0946.exe, after enconding them to the Speex audio format using VBR, as follows: speexenc.exe --vbr <input-wav-file> <output-spx-file> The output of speexenc.exe --version* *is: speexenc (Speex encoder) version speex-1.2beta3 (compiled Dec 11

hi are there any codecs around that allows high quality as in "studio lite"? it may consume high bandwidth, and hopefully allow some packet loss. roy

----- Original Message ----- From: "Alexander Chemeris" <Alexander.Chemeris at sipez.com> To: "Vincent Burel" <vincent.burel at vb-audio.com> Cc: "Conrad Parker" <conrad at metadecks.org>; <speex-dev at xiph.org>; "Jean-Marc Valin" <jean-marc.valin at usherbrooke.ca> Sent: Thursday, November 13, 2008 11:31 PM Subject: Re: