search for: 24khz

...Dial, s, 4) exited non-zero on 'SIP/ng-00000000' core show translations doesn't include any SILK. SILK is installed: core show codec 100018 100018 SILK Custom Format 8khz 100018 SILK Custom Format 12khz 100018 SILK Custom Format 16khz 100018 SILK Custom Format 24khz sean

...patent jungle. The only comment I've seen about use of Opus for audiobooks was jmvalin saying in response to someone on his blog that Opus's ability to do fullband would be a key advantage here. This seems kind of counterintuitive to me- can people even ABX human speech at a 32 or even 24kHz sample rate from speech at 48kHz, much less hear a large quality difference? A number of audiobooks I've listened to have used 22kHz mp3s without being clearly objectionable, and in my personal use I've had decent results using the -voice LAME setting (downsamples to 32kHz and encodes a...

I have been trying some different sample rate and bitrate combinations to get a feel for how CELT behaves. I then encoded it with a couple of different frame sizes and sample rates. There were some small differences, but nothing horrible. So, I decided to run with 24KHz sample rate with 16 bit samples. This seemed like a reasonable tradeoff against the quality degradation that my folks could perceive. Then I tried some bitstream rate changes 128Kbps, 64Kbps, and 32Kbps. 128K and 64K are fine. 32K is *terrible*. Really, really horrendous. Is this expected...

...00ms) 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 1150 1000ms 1296ms + 193ms = 1507ms FIXED 8kHz 2 1150 1000ms 1014ms + 147ms = 1181ms FIXED 8kHz 1 1150 1000ms 1086ms + 135ms = 1241ms FIXED 8kHz 1...

Hi Im using Icecast2 and SAM2 to broadcast a 56kb 24khz Stereo 2 Channel 6.Kb MP3 stream on a 512k ASDL connection. I know that the number of listeners will be LIMITED to 3 or so - is there any way I can actually calculate this ? I passed my broadcast link to a friend yesterday and they told me that they recieved a 'Server Full' error, is ther...

...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. (However; If my assumptions are correct there could be a possibility to make my I2S device master and, thus, be able to pull the data at 16kHz) Consequently I have been looking at “opus custom” as a solution but even with a (seemi...

...r 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

...with 11.025kHz. Note that these aren't the LAME default sampling rates for these bit rates. If you want to go 40kbps, some quick informal testing gives the following results: Stereo: LAME default is 16kHz, but to my ears, you can get acceptable results at 22.05kHz. Mono. LAME default is 24kHz, but 32kHz sounds just fine. 44.1kHz is starting to push it, but since it gets rolled off anyway, there's really no point in going that high. At 32kbps, LAME's defaults are 16kHz stereo and 22.05kHz mono, and these are what I'd probably recommend. This is just from some quick info...

a comparison between the two formats can be read at http://www.digit-life.com/articles/oggvslame/index.html . i first heard of this from Vorbis Extreme ( http://solair.eunet.yu/~aldov/ ). anyone see any problems with the review? -- noodlez: Karol Pietrzak GPG/PGP-KeyID: 0x3A1446A0 --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To

Hello, I've got two streams, one for broadband, one for dialup. Well, having had occation to use a dialup connection recently i checked the dialup stream. Although it was streaming what the broadband stream was, the audio quality was audibly worse. It didn't buffer, but it didn't sound as clear as the broadband stream. I used lame to encode the tracks to mp3 and used it's

Hi. I've been browsing the archive on this topic and only found a few notes, all dating from a year ago (almost too precisely :) ) -- hope I haven't skipped the mails on that matter, sorry if I did. <p>Basically, I will get in the next few months a MOTU 896, that have 8 i/os in 24/96. I do pro sound recording, so it's more or less my business to have such a piece. Of

...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. (However; If my assumptions are correct there could be a >> possibility to make my I2S device master and, thus, be able to pull the >> data at 16kHz) >> >> >> >> Consequently I have been l...

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I've just installed RC2 and I'm very excited about the quality. It's so much better than MP3. This is the first version I've used since I just found out about Ogg Vorbis. I did notice that very high frequencies seem to be missing but since not many people can hear much above 18 KHz it's not much of an issue. I suppose this resolves the hiss problem so prevalent in MP3.