search for: 192khz

...> Hi all, > > > > i'd like to ask what the best options are for safe 24bit 96kHz > > archiving. Currently i'm only using -8 but there are also some other > > options like block size etc. Can anyone suggest? > > I haven't heard of anyone using FLAC for 192khz, so you'll > probably just have to experiment. when you have some results, > let me know so I can tune the defaults better. > > for high sample rates, tweaking -b and -r (and to a lesser extent > -l) is important > > for high resolutions (24bit), -l, -e, -p, -q are relev...

Hi all, i'd like to ask what the best options are for safe 24bit 96kHz archiving. Currently i'm only using -8 but there are also some other options like block size etc. Can anyone suggest? Also i'd like to ask whether a no-lax 192kHz mode is planned in flac, seems like 192kHz isn't directly supported although flac can compress such rate in lax mode. My last question - is there any chance for flac to support w64 files? That would be cool since it would allow to compress >2Gb files. (not sure if flac can compress >2Gb...

Greetings, In reviewing the changelogs it?s unclear in which release FLAC began supporting a sample rate of 192kHz. The reason for my question is that there are many forums and university studies that state that FLAC does not support a sample rate of 192kHz however the current documentation (assumed 1.2.1b) under FORMAT under FRAME_HEADER does note that it is supported. If it was not supported from the...

...<marpet@naex.sk> wrote: > Hi all, > > i'd like to ask what the best options are for safe 24bit 96kHz > archiving. Currently i'm only using -8 but there are also some other > options like block size etc. Can anyone suggest? I haven't heard of anyone using FLAC for 192khz, so you'll probably just have to experiment. when you have some results, let me know so I can tune the defaults better. for high sample rates, tweaking -b and -r (and to a lesser extent -l) is important for high resolutions (24bit), -l, -e, -p, -q are relevant. > Also i'd like to ask...

That article is a bit too dismissive. I agree that one cannot hear the difference between 48KHz/16bit and 192KHz/24bit if you just transfer the data directly to the audio output device. As such, there is no good reason for Opus to support higher than 48KHz (especially since this is lossy compression, anyway). However, in general, that's not all you do with audio data. 192KHz is useful for the same reas...

Hi!, I wondering if It's possible to change 48khz sample rate limit?, I'm Planing to encode with OGG codec a audio signal but I need that OGG Encoder works with 192khz of sample rate. It's Possible? Any Suggestions?

...t xiph.org" <opus at xiph.org> > Message-ID: > <1350388129.15298.YahooMailNeo at web171605.mail.ir2.yahoo.com> > Content-Type: text/plain; charset="iso-8859-1" > > Hi , I have read that it is posible to encode higher sample > rates like 96 khz or 192khz? and the output is 48 khz, > the resample is internally.? http://wiki.xiph.org/OpusFAQ > But it is possible to encode? 44.1khz. It is resampled to > > 48khz or I have to make the resample by myself and > > then encode it with opus. > > thnx, arctor > -------------- next p...

.... You can do as much linear filtering as you like, the > stuff above 20 kHz isn't going to help you. But lots of effects are not linear--simulating a tube guitar amplifier, for example. Even something as straightforward as resampling a signal to 44.1KHz is going to benefit from starting at 192KHz rather than 48KHz. There may not be more signal information but there will be less noise. -a -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 538 bytes Desc: OpenPGP digital signature Url : http://lists.xiph.o...

Hi , I have read that it is posible to encode higher sample rates like 96 khz or 192khz? and the output is 48 khz, the resample is internally.? http://wiki.xiph.org/OpusFAQ But it is possible to encode? 44.1khz. It is resampled to 48khz or I have to make the resample by myself and then encode it with opus. thnx, arctor -------------- next part -------------- An HTML attachment wa...

Hello, I'm an audiphile. Will opus ever be developed to work with sampling rates higher than 48kHz? (I hope so)

...ependent out-of-call messaging support. Text messages not associated with an active call can now be routed through the Asterisk dialplan. SIP and XMPP are supported so far. * New highly optimized and customizable ConfBridge application capable of mixing audio at sample rates ranging from 8kHz-192kHz (More information available at https://wiki.asterisk.org/wiki/display/AST/ConfBridge+10 ) * Addition of video_mode option in confbridge.conf to provide basic video conferencing in the ConfBridge() dialplan application. * Support for defining hints has been added to pbx_lua. * Replacement of...

...ependent out-of-call messaging support. Text messages not associated with an active call can now be routed through the Asterisk dialplan. SIP and XMPP are supported so far. * New highly optimized and customizable ConfBridge application capable of mixing audio at sample rates ranging from 8kHz-192kHz (More information available at https://wiki.asterisk.org/wiki/display/AST/ConfBridge+10 ) * Addition of video_mode option in confbridge.conf to provide basic video conferencing in the ConfBridge() dialplan application. * Support for defining hints has been added to pbx_lua. * Replacement of...

...t wavpak is the only compressed container format that supports the 705/768 sample rates but it is very poorly supported and extremely slow during decompression, and fully uncompressed WAV data is extremely cumbersome. Most DACs do appear to turn off their anti-aliasing filters at sample rates above 192kHz so 384 may well be the limit of any theoretical benefit, but many DACs now support input up to 768kHz (a very small number support even higher.) Studio ADCs often record in the Mhz range, and although they apparently always decimate as they sample, there may be a theoretical use for the data as is....

Hi there - I am currently in the middle of a project where I need to configure an Icecast Server and I am using Ices as the encoder. I want to configure Ices to take in a stream with a sample rate of 128 kHz and output it to the Icecast server. My question is, what are the minimum and maximum sample rates that Ices will continue to work with? Thank you, Heidi Young Software

...und, but if I can pay $50 and get something that sounds pretty close, I'd rather do that, just because computer technology changes faster than audio technology --- I'm unlikely to have the same card in 5 years, whereas my stereo should be good for 20 (assuming, as seems likely, that 192KHz, 24 bit audio _isn't_ around the corner after all!) * Stereo is necessary, 5.1 isn't. This is for music, not movies. But if there's a great 5.1 sound card, I'll take that. If you email me directly, I'll summarise to the list. I'd also very much appreciate it if you...

...d environment. I have it fully up and running but I am trying to optimise the performance vs. the power it takes to decode the stream. Using the reference decoder code with a few optimisations for the hw I'm on I experience quite a difference in mips depending on the level. For instance, on a 192kHz/24 bit file the difference is almost that it takes twice as many mips on level 8 vs. level 1. Unfortunately I can't tell our customer that they need to decode at a certain level, BUT I want to minimize the frequency the processor is running on. Since I have no problem reading a 441/16 from a...

...svn/asterisk?view=revision&revision=314598 If you are already familiar with ConfBridge from Asterisk 1.6.X and 1.8, forget everything you know. This is a completely revamped, highly optimized, and feature rich conferencing application capable of mixing sample rates from 8khz all the way up to 192khz! Exciting right?! So Go! use it, test it, and report back! Tell us what you like, what you don't like, if you want a feature that doesn't yet exist, and report bugs. This conferencing application has huge potential and we need community feedback. Asterisk 1.10 isn't that far away,...

...me PLL source, presumably for >simplicity. >> >> 2. if it is a AC97 or HD sound card, clock frequencies of their hardware >> ADC and DAC are all fixed to 48/196KHz. Then why their are still not >> synchronised? >> >Not really. They are *approx* 48kHz or *approx* 192kHz. >> >> > It seems like Skype's speakerphone works OK on these machines, so maybe >> > they do something tricky to measure the slip rate and resynchronise in >> > software. On the other hand, they might just fall back to a crude echo >> > suppression sch...

...ing to optimise the performance > > vs. the power it takes to decode the stream. > > > > Using the reference decoder code with a few optimisations for the > > hw I'm on I experience quite a difference in mips depending on the > > level. > > For instance, on a 192kHz/24 bit file the difference is almost that > > it takes twice as many mips on level 8 vs. level 1. > > > > Unfortunately I can't tell our customer that they need to decode at > > a certain level, BUT I want to minimize the frequency the processor > > is running on....

...compressed container > format that supports the 705/768 sample rates but it is very poorly > supported and extremely slow during decompression, and fully > uncompressed WAV data is extremely cumbersome. Most DACs do appear to > turn off their anti-aliasing filters at sample rates above 192kHz so > 384 may well be the limit of any theoretical benefit, but many DACs > now support input up to 768kHz (a very small number support even > higher.) Studio ADCs often record in the Mhz range, and although they > apparently always decimate as they sample, there may be a theoretical &gt...