similar to: Ogg/Vorbis Downsampling?

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

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

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, 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,

Try this with Samba 2.2.x with a Win2K box ... net use x: \\sambaserver\sharepoint x: mkdir test cd test mkdir test The second "mkdir test" will fail with the following error ... "A subdirectory or file test already exists." Any attempt to make the directory, even with an explicit path, as in ... mkdir test\test mkdir x:\test\test ... results in the same error. It

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

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

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 guys, Another strange behavior with some test servers. They are booted using 2.6.11.12, xen-2.0.6, Debian. The only daemons running on dom0 are : xend, smtp, quagga (ospf), portmap (nfs client), sshd. Memory usage at startup is aroung 30 MB. Dom0 reserved memory is 256 MB. The only things that happen on this server are domU''s start/stop (xm create, xm shutdown) and a

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

We have DELL 2650 server and I found recently /var/log/messages have following messages continue pop-up: =========================================== Dec 3 15:47:06 HOST1 kernel: Free pages: 17816kB (1664kB HighMem) Dec 3 15:47:06 HOST1 kernel: Active:7463815 inactive:656000 dirty:0 writeback:0 unstable:0 free:4454 slab:119979 mapped:730440 pagetables:63277 Dec 3 15:47:06 HOST1 kernel:

Hey everyone, we finally bit the bullet and threw out our firstborn for public consumption. For those not yet in the know, Flumotion is a streaming media server based on GStreamer and Twisted under development by Fluendo. This release gives users access to the basic features of the server and demonstrates its distributed capabilities. We will follow up with a release within a couple of weeks to

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

Hi All, I am trying to get speex working on the Mac and am running into issues. I got the examples working, but am now trying to make speex, which expects 8000 Hz and 160 samples per buffer (320 bytes per buffer), work with the Mac's built-in audio recording, which uses either 11025, 22050, or 44100 Hz and 1024 samples per buffer (2048 bytes per buffer). I just need to know if

Hi, I'm experimenting with IceCast2, using DarkIce to generate the stream. I have found some peculiarities with the vorbis bitrates. In DarkIce, I call vorbis_encode_init() with about the following values: vorbis_encode_init( &vorbisInfo, 2, 44100, 96, 96, 96); which by all reasons should generate a 96 kb/s stream, as all max_bitrate, nominal_bitrate and min_bitrate are set to 96.

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

I have a partly damaged CD, from which I extracted 192Kb/s Vorbis files a while ago (before it got damaged) and I'd like to get FLAC files. `cdparanoia' extracted most of the data from the CD, but there are some holes due to the damages. How could I fill those holes in my FLAC files using the Vorbis files? Stefan

I have a partly damaged CD, from which I extracted 192Kb/s Vorbis files a while ago (before it got damaged) and I'd like to get FLAC files. `cdparanoia' extracted most of the data from the CD, but there are some holes due to the damages. How could I fill those holes in my FLAC files using the Vorbis files? Stefan