search for: 22khz

I've just finished a little encoding set and the following happened: 22Khz resampled, q-0.4 encode = ~38kbs = one error at the beginning only. 22Khz resampled, q-0.5 encode = ~37kbs = 6 to 7 sleep errors in the first second or so, nothing after that. 22Khz resampled, q-0.6 encode = ~35kbs = 14 to 15 sleep errors then nothing. 22Khz resampled, q-0.8 encode = ~33kbs = Ma...

> I agree with you here ... I wish that I could put the stream > into 30kbps/22khz stereo but the little Pen 166 MMX just > can't handle that. 30kbps/22khz stereo, which I think would only be possible with managed bitrates, would sound worse than mono. At these bitrates, stereo is luxury you cannot afford. I think most listeners would not even notice they were listeni...

hello all! After having read http://www.xiph.org/ogg/vorbis/listen.html and listened to the examples on that very educational page, I decided to augment the info. This example is more simple, it involves the comparision between two streams of the same radio station, FranceInter (a station of Radio France in Paris). The ogg stream is running at around 30 kbps/11 kHz in stereo. The WMA stream

Boink wrote: > They're streaming only in 22050/11khz mono. > I'm doing my stream around 30 kbps/11khz in *stereo*. Just my opinion but I've found streaming with quality -1 22khz Mono produces the best sound quality for 32kb/s. 11khz stereo sounds like crap in comparison. I'm just waiting (impatiently) for OddCastDSP(v1) to support 22050 mono via the SQRSoft crossfader. If OddSock is too busy, perhaps someone else could take a look at the source. :-) Ross. --- >8...

Hi, Is it possible to generate an ogg stream at less than 22KBps ? I tried several configuration with ices and darkice, but I always get an configure encoder error: [2003-03-17 16:07:17] INFO input-oss/oss_open_module Opened audio device /dev/dsp at 2 channel(s), 44100 Hz [2003-03-17 16:07:17] INFO audio/resample_initialise Initialised resampler for 2 channels, from 44100 Hz to 22050 Hz

...:19:18AM -0800, HJ wrote: > > --- boink <boink@tetter.xs4all.nl> wrote: > > The ogg stream is running at around 30 kbps/11 kHz in stereo. The > > WMA > > stream (provided by yacast.fr) is running at 16 kbps/22 kHz in > > mono. > > Is stereo vs. mono, and 22khz vs. 11khz even a fair comparison? > > 22khz vs. 11khz: 22khz, by definition, gives the 22khz stream much > more frequency range and *opportunity* to sound clearer. Yes, there > are more audible artifacts in the .wma than in the Vorbis stream, > but IMHO the sampling rate should mat...

On 2002.12.02 16:08 SwiftBiscuit wrote: > I've just finished a little encoding set and the > following happened: > > 22Khz resampled, q-0.4 encode = ~38kbs = one error at > the beginning only. > > 22Khz resampled, q-0.5 encode = ~37kbs = 6 to 7 sleep > errors in the first second or so, nothing after that. > > 22Khz resampled, q-0.6 encode = ~35kbs = 14 to 15 > sleep errors then nothing. > &gt...

--- boink <boink@tetter.xs4all.nl> wrote: > The ogg stream is running at around 30 kbps/11 kHz in stereo. The > WMA > stream (provided by yacast.fr) is running at 16 kbps/22 kHz in > mono. Is stereo vs. mono, and 22khz vs. 11khz even a fair comparison? 22khz vs. 11khz: 22khz, by definition, gives the 22khz stream much more frequency range and *opportunity* to sound clearer. Yes, there are more audible artifacts in the .wma than in the Vorbis stream, but IMHO the sampling rate should match anyway. Stereo vs. mo...

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 sugge...

...he delay between server and client grows as listening is going on. As pointed out Philipp yesterday to me, it's approximately a 5% drift, apparently caused by the client audio card clock. But, then since there are no cuts in the sound as I am listening, it means the audio rate is lower than the 22kHz of the files, right ? And a 5% drift at 22kHz is around 1kHz difference ! And even if I did not properly measured this behavior, I am not subjectively able to hear a difference in the output. So, what are the solutions for locking clocks so that the latency is constant ? Sorry if I don't unde...

...t? As others have said, 16kbps should do the trick. Keep in mind though that the quality of the sound will also depend on the sampling rate. MP3 will handle some higher sampling rates higher than some of the lame defaults. For example, 16kbps 11khz mono sounds very reasonable. At 24kbps, either 22khz mono or 11khz stereo sound pretty good, though the latter is a bit iffy under lame. At least with 24kbps however, these aren't the lame defaults. Mono will give you 16khz and stereo gives you a lowly 8khz (yuck!). You can get lame to do higher sampling rates by using the resample argument ra...

I'm using 22khz wavs because 16khz doesn't provide the quality i need in the original wavs. I can't get 32khz wavs because that part of the project is finished and not redoable. The .spx's don't have to be as high quality, so i guess i could convert them to 16khz. I was trying 17kbps, ultra-wideb...

...st, SPEEX_GET_LOOKAHEAD, &lookahead); >> ... >> nb_encoded = -lookahead; >> >> Can someone explain what this means? > > The lookahead is the number of samples you need to discard at the > start. > > Jean-Marc In my application, even 5ms (110 samples at 22KHz) matters. So what should I do to avoid discarding samples at the beginning? 1. Turning off look ahead? 2. Padding 0's at the beginning.

...each server, e.g. DarkIce could acquire > the audio in stereo and send it to a server in mono and in stereo to > another server, which is AFAICT impossible today. I agree! Also, something I've been looking for is a way to pull sound from the dsp device at 44kHz and then downsample it to 22kHz for one of my two streams. Ideas? DarkIce sure does rule. -samuel --- >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-request@xiph.org' containing only the word 'unsubs...

...er, and how can I use it to solve my problems? Specifically: 1) Assuming lossless, in-order, but highly irregular delivery of packets (as I'm witnessing), what advantage does the jitter buffer offer over going straight to the Speex decoder? 2) Assuming samples arrive at an average rate of 22KHz, but arrive in a highly irregular fashion, is there any way to ensure regular playback other than to just wait some "prebuffer" duration before beginning playback? How do I pick the smallest prebuffer duration to accomodate a given connection's jitter? 3) Assuming I want to deli...

...ture audio (1 channel, 22050Hz, 16 bits/sample) in blocks of the frame size. However, the encoder lags significantly - typically upwards of 2000 frames, at complexity <= 1 and quality <= 3. I've a On the decode side, the decoder is not able to do real time at these sampling rates (16kHz, 22kHz and 32kHz). The only sampling rate that seems to work at real time is 8kHz for decode. However, encode at 8kHz is still nowhere near real time. Is there anything I can do to get to the higher sampling rates and real time? Or is this just not possible? FWIW, the basic app uses MFC and the encode a...

Thanks for offering to help, Karl. Config - I thought my email was long even without the config files so I didn't include them! They are normal random playlist configs. What aspects are you looking for? CPU - vmstat 1 gives consistant CPU readings of around 2,1,97 for us,sy,id respectively when no one is listening to the streams. There's hardly any change when a client starts listening.

...ram and an 100MB ethernet nic(possible a good system). If i want to start more than 3 streams , all streams are running not continously. many dropouts and squelches while playing. is there an config problem nor is the hardware too small ? the stream running with 128kbs/44khz. ive tested with 64kps/22khz and 32kps/22khz and there is no better run of the server..... if there is an need for my config-files i will post them ... hope for some answers .... thx nils from germany --- >8 ---- List archives: http://www.xiph.org/archives/ icecast project homepage: http://www.icecast.org/ To unsubscri...

Hello, To start with, thanks a lot for making such a great voice codec available! Having recently upgrading to speex rc1, It occurred to us that there seems to have been a regression in the quality of encoding since version beta1. We are compressing some 22khz wave files in wb mode with maximum quality / complexity in VBR, and the result was really great with speex beta1. With rc1 (or beta3), there is a clear degradation for fricatives, which gives a very audible (and annoying) feeling of a muffled voice. This problem does not seem to affect CBR encodin...

...t 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 as 56kbps abr). The recent hydrogenaudio tests showed Opus CELT modes trumping the best of breed high-latency codecs at 64kbps despite ha...