search for: downsample

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 hi...

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

...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 introduce aliasing but I found no good highquality solution to this :( The question is, is this FIR-decimator the best method to downsample? I thought of converting the frame to the frequency domain, drop components > 8khz and convert back, but I'm not sure if that would produce a better signal or worse. Also, would it be best to do echo cancellation and preprocessing at 48khz and then decimate to 16khz, or (as I currently...

...port 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 data as needed. I didn't find such possibility for vorbis. For lame, this increases the processing need quite a lot (almost double as I measure). 2. Downsample in DarkIce, by simple means. For example, from 44.1kHz -> 22.05kHz, simply send one sample for every two imput samples, taking...

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

...audio programs. > > However, I still downmix to mono by just averaging the left and right > samples, which (from my googling) I can tell will introduce aliasing but I > found no good highquality solution to this :( > > The question is, is this FIR-decimator the best method to downsample? I > thought of converting the frame to the frequency domain, drop components > > 8khz and convert back, but I'm not sure if that would produce a > better signal or worse. > > Also, would it be best to do echo cancellation and preprocessing at 48khz > and then decimate...

...oblem 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. It's talk, and I'd like to squeeze 200 connections into a 5Mbps pipe. I'm currently able to do a 24K mono stream, but 11K would do the trick. I've seen references to <downsample>, but can't find any hard documentation on it (other than a couple sample con...

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

...n 2009 Jul 2009 NA NA NA NA 3 > points(t2.monthly, type="p", col="blue") > lines(na.locf(t2.monthly), col="blue") # as an example of why I might want to do this. > Similarly, it would be nice if one could conveniently downsample a time series, choosing to keep only the Nth point, or the sum or the average of the previous N points, etc... I can see how that particular application could probably be accomplished relatively easily using rapply and a subsetting operation. However it might be nice to have a convenient wrapper f...

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

...*/ #endif /* !defined(FIXED_POINT) && defined(HAVE_ARM_NE10) */ diff --git a/celt/celt_decoder.c b/celt/celt_decoder.c index 4304a3e..304f334 100644 --- a/celt/celt_decoder.c +++ b/celt/celt_decoder.c @@ -278,8 +278,9 @@ void deemphasis(celt_sig *in[], opus_val16 *pcm, int N, int C, int downsample, c static #endif void celt_synthesis(const CELTMode *mode, celt_norm *X, celt_sig * out_syn[], - opus_val16 *oldBandE, int start, int effEnd, int C, int CC, int isTransient, - int LM, int downsample, int silence) + opus_val16 *oldBandE, int start, int effEnd, int C,...

...*/ #endif /* !defined(FIXED_POINT) && defined(HAVE_ARM_NE10) */ diff --git a/celt/celt_decoder.c b/celt/celt_decoder.c index 4304a3e..304f334 100644 --- a/celt/celt_decoder.c +++ b/celt/celt_decoder.c @@ -278,8 +278,9 @@ void deemphasis(celt_sig *in[], opus_val16 *pcm, int N, int C, int downsample, c static #endif void celt_synthesis(const CELTMode *mode, celt_norm *X, celt_sig * out_syn[], - opus_val16 *oldBandE, int start, int effEnd, int C, int CC, int isTransient, - int LM, int downsample, int silence) + opus_val16 *oldBandE, int start, int effEnd, int C,...

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

...oding 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 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...

We want to record inbound channels at 16kHz, but send only 8kHz to our peers. I've set our default profile in sip.conf to disallow all but g722, and the peers disallow all but ulaw. We have a proxy in front of Asterisk that is configured to disallow all but G722 also. My test calls show inbound to the proxy is recorded at 16kHz, inbound in Asterisk is only 8kHz, and the peers receive 8kHz. So