search for: downsampling

...med ch_wave. `sox' performs terribly, so I didn't use it. However, I heard some unpleasant sound at relatively high frequencies in the *downsampled WAV file*, but not in the encoded OGG file. So I assume I need some even better downsamplers. Is there any on Linux? P.S. Why not put such downsampling things in the encoder, activated when the quality setting is really low? Or better, save bits by some way cleverer than downsampling? <p><p><p><p><p><p><p><p>__________________________________________ È«³¡ÌؼۣºÊýÂëÏà»ú¡¢ÕÆÉϵçÄÔ¡¢MP3 http://shoppin...

Hi! We want to over our stream in better quality (128 or 256) - but we still have listeners using ISDN ... what's the best way to create a 56'er stream from the 128er send to the server? The downsampling has to run on the debian streaming server. Greetings from Germany Philipp

Dear Vorbis Gurus to whom I owe a debt of gratitude for creating such a kick-ass audio encoding scheme: I'm probably using the wrong term of "downsampling" here, but here goes. I remember reading about Vorbis being designed with streaming in mind. I was wondering if one aspect was to allow easy ad-hoc downsampling (e.g. going from 192kb/s to 128kb/s) without re-encoding. Does such a facility exist in the Ogg format or Vorbis encoding? Regards,...

...limited to the internal sampling frequency of the sound card, which for my Audigy 2 is 48 khz, and only delivers stereo. As I understand it, just picking every 3rd sample and averaging left and right will introduce aliasing, so I added a FIR-filter to remove all components above 7.5 khz before downsampling. While I can't say I really notice any difference in audible audio quality, the samples seem to have slightly better SNR when inspected in various audio programs. However, I still downmix to mono by just averaging the left and right samples, which (from my googling) I can tell will introduc...

Hi, Maybe a bit off topic for this list, bt anyway. I have received several feature requests for DarkIce to support downsampling of the audio input before passing it to lame or ogg vorbis. For example the audio read from the soundcard would be 44.1kHz, and lame would get it at 22.05kHz. I figure two ways of doing this: 1. For lame, one can specify the input and the desired mp3 sampling rate, and lame will downsample the da...

Jack Moffitt wrote: > This isn't good enough. Just rip out lame's downsampling code (or > sox's) and use that (as long as your also under a compatible license). DarkIce is GPL, should be OK. > Those are both pretty good. Averaging every two samples just doesn't > cut it :) I thought so :( But I wouldn't need to rip out the code, if vorbis supported...

Hi, I am looking for ways to donwsample one-dimensional vectors. For example, x=sample(1:5, 115, replace=TRUE) How do I downsample this vector to 100 entries? Are there any R functions or packages that provide such functionality. I did find the zoo package and the aggregate() function, but these appear to be rather specific for time-series. Thanks in advance, Jan

...rnal sampling frequency of the > sound card, which for my Audigy 2 is 48 khz, and only delivers stereo. As > I understand it, just picking every 3rd sample and averaging left and > right will introduce aliasing, so I added a FIR-filter to remove all > components above 7.5 khz before downsampling. While I can't say I really > notice any difference in audible audio quality, the samples seem to have > slightly better SNR when inspected in various audio programs. > > However, I still downmix to mono by just averaging the left and right > samples, which (from my googling)...

On Mon, 2003-08-25 at 17:04, W. Kevin Pedigo wrote: > But if your problem is serving more bandwidth than you've got, you gotta > serve less (narrower or fewer streams) or get more bandwidth. It's that > simple. Tell us what you want to do about it, and we'll try to help. OK. I've gotten everything running with one problem. I'd like to downsample a live stream.

Hello, We're streaming radio programs at both 128kbsp and 32kbps, but only archiving the 128kbps stream to save storage space. I'd like to give users a similar choice of bitrates when they request an archived stream (served through icecast's /file/ functionality). Is there a way to change the bitrate on the fly, or do I really need to save archive both bitrates? Thanks for the

As the subject implies, my question is: is it possible to use the -b (or -M) flag at non-44K sample rates? I'm working with an application that is trying to optimize for very small audio filesize. I found that downsampling to 11K and then using q0 gives high compression, but won't seem to drop below 64kbps or so. It seems like the combination of downsampling, then reducing to 30kps would compress things even further, but I'm not sure if this is allowed. Any ideas? <p>--- >8 ---- List archives:...

Hi I have data that is sampled (in time) with a certain frequency and I would like to express this time series as a time series of a higher (or lower) frequency with the newly added time points being filled in with NA, 0, or perhaps interpolated. My data might be regularly or irregularly spaced. For example, I might have quarterly data that I would like to handle as a monthly time series with

https://bugzilla.samba.org/show_bug.cgi?id=11075 Bug ID: 11075 Summary: Shouldn't --inplace fail immediately if it can't make files? Product: rsync Version: 3.1.0 Hardware: All OS: All Status: NEW Severity: normal Priority: P5 Component: core

Can anyone recommend a good library for performing audio downsampling? I intend to start playing around with "libresample" (http://ccrma-www.stanford.edu/~jos/resample/README-libresample-0.1.3.txt), as well as taking a look at "Secret Rabbit Code" (http://www.mega-nerd.com/SRC/), but I'd love some opinions before I get too involved with eit...

> > I am talking about the original sweep. > > The original sweep stops pretty close to 24 kHz. I mean the original sweep _as_encoded_, sorry.

Optimize opus decode (float only) use case using ARM NE10. Mainly effects opus_ifft and ctl_mdct_backward and related functions. Work based on previous Encode optimization using ARM NE10 library. TBD: Add commit id of upstream Encode NE10 optimization patch so that users have reference of how to enable this optimization Signed-off-by: Viswanath Puttagunta <viswanath.puttagunta at

...alling, not rising. Finally, the fact that first the undesired frequency > comes in pretty quickly and then the original frequency disappears very > quickly; so at one point, they're both there (and this point happens to be > at 8 kHz). > > > What makes you think there is any downsampling involved? > > The fact that frequencies higher than 20 kHz are not filtered out but are > rather replaced with frequencies higher than 4 kHz. Well let's hear it from the horse's mouth, if there are opus developers reading this list: given a 48 kHz wav file, does opusenc downsampl...

Optimize opus decode (float only) use case using ARM NE10. Mainly effects opus_ifft and ctl_mdct_backward and related functions. Work based on previous Encode optimization using ARM NE10 library. TBD: Add commit id of upstream Encode NE10 optimization patch so that users have reference of how to enable this optimization Signed-off-by: Viswanath Puttagunta <viswanath.puttagunta at

Am 04.02.2015 um 12:31 schrieb Timothy B. Terriberry: > M. Pabis wrote: >> 1. Each thread deals with frames from intra frame up to next intra frame >> - 1; > > This works if you know where the intra frames are. Could this information be gathered by having one thread encode a downsampled version of the input video sequence, or would this be a bad predictor? Who knows,

Hello All, I extended the libNE10 optimizations for float towards mdct_backwards/opus_ifft. I am able to get about 14.26% improvement for Decode use case now on my Beaglebone Black. Please see [1] for measurements. Questions 1. Since this patch needs to go in after Encode [2] patch) should I submit this as patch series? 2. Since Jonathan Lennox posted intrinsics cleanup [3] patch, should