similar to: Downsampling

Ok, this is slightly offtopic, but relates to the quality of input for speex :) I'm working on echo cancellation by means of sampling the wave mix of the sound card as well as the microphone. I originally had two sound cards, which had some synchronization problems (now solved, more or less), but I have also discovered a much better solution using ASIO 2.0, which enables me to sample

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,

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

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

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, For transforming stereo to mono, averaging is fine and that's what everybody does. For sampling rate conversion, it's another matter (too long for this email) and you should read a bit about it a perhaps grab a library that does that. As for echo cancellation, it will be less complex (and as good) on a (cleanly) down-sampled signal (and certainly not on stereo). Jean-Marc Le mardi

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

Hi Jean-Marc, didn't get a reply to my last post (see below) -- do you have no idea what happens here? After some more tests, I disabled the DIV32_16 Blackfin optimizations and now get good quality on the Blackfin. But when I have overdrive on the input, things become very bad -- I'm not sure if this is really a filter stability issue like I wrote some weeks ago. I use the speex

Hi, I just started to examine the DIV32_16 function (Blackfin ASM version), and wondered why the return value of the function inside 'fixed_bfin.h' is of type 'spx_word16_t', but the local variable 'res' which is returned by this function is of type 'spx_word32_t'. Is this a trick of optimization or a bug? (Same question for PDIV32_16 and MAX16, too!) best

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.

Hi, I tried to figure out what the problem is -- but it seems to be totally different from what I expected. My status at the moment is: - computing results for "generic" and "Blackfin ASM" versions of the DIV32_16 function are the same, there is no "algorithmic bug" - Instead, there seems some sort of memory corruption: When I comment out the DIV32_16 function

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

On Aug 07 22:04:21, petrparizek2000 at yahoo.com wrote: > > The encoded opus file is 48kHz, > > so how would the output wav be resampled from 16kHz? To be clear: did you mean the opus output of opusenc or the wav output of opusdec? > > What are those "clear signs" exactly? > > The things that I can hear while listening at 1/2 or even 1/4 of the > original

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,

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

On Fri, 16 Nov 2001, Jerome Alet wrote: > one thing that would be nice in DarkIce would be to allow the user to pass > specific reencoding options for 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

On Aug 07 08:30:31, hans at stare.cz wrote: > On Aug 07 00:41:52, petrparizek2000 at yahoo.com wrote: > > ????#1. To test encoding at low bitrates, I encoded a sine sweep at 12 kbps > > with Opusenc and then decoded the resulting file with Opusdec. > 1) Opusenc --bitrate 12 --downmix-mono Sweep50.wav Sweep50.opus Why are you using a stereo file containing the same sweep in both

> Why are you using a stereo file > containing the same sweep in both channels > and then downmixing to mono? When I first tried encoding at a higher bitrate, I needed to test the different behavior of the "mid" (l+r) and "side" (l-r) channels. That's why I made the first sweep identical on both the left and the right channel (i.e. "side" is silent)

On Aug 07 00:41:52, petrparizek2000 at yahoo.com wrote: > ????#1. To test encoding at low bitrates, I encoded a sine sweep at 12 kbps > with Opusenc and then decoded the resulting file with Opusdec. What sine sweep exactly? How did you obtain it, and how exactly did you encode and decode it? Jan > The strange > thing was that even though the output wave file was at 48 kHz, it