search for: downsampled

It is commonly said here that if I want to make AM radio-quality stuff at very low bitrates, a good way is to downsample. I downsampled a song to 11025Hz mono and encoded with -q 0, the result is about 18kbps and is at least radio quality. The downsampler I used is from Edinburgh speech tools, named ch_wave. `sox' performs terribly, so I didn't use it. However, I heard some unpleasant sound at relatively high frequencies...

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

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,

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

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

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

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 either. Free would be best, but

> > 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

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

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, perhaps one can also gather data on relative bitrate distribution between segments. Maik

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