similar to: -b flag at low sample rates?

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,

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

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

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! 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

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

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

Hi I want to encode at 64kbps at a 22050 rate ... this is not possible ... why? For me it seems a waste of cpu to encode at 64kbps with 44100 sample rate (of course I might be wrong :). I saw in the vorbisenc.c that aprox_bitrate_to_vbr returns -1 to every sample rate that is not > 42000 ... is this safe to change ? Thanks, Nicu. <p>--- >8 ---- List archives:

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

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'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.

I've just done some rudimentary testing to see how Vorbis degrades at absurdly low bitrates without downsampling. In summary, don't hope for anything decent below -q 0 for now. I tried oggenc -b <bitrate> -M <bitrate> for the below and a few in between: 24k - spectral energy "floor" captured decently, but many pure-tone blips (think old computer movie sound effects)

On 03/02/2018 11:29 PM, Ilia Mirkin wrote: > On Fri, Mar 2, 2018 at 5:16 PM, Mario Kleiner > <mario.kleiner.de at gmail.com> wrote: >> On 03/01/2018 07:21 PM, nouveau-request at lists.freedesktop.org wrote: >>> >>> >>> Message: 1 >>> Date: Thu, 1 Mar 2018 08:15:55 -0500 >>> From: Ilia Mirkin <imirkin at alum.mit.edu> >>>

Hi Nuno, Chandler, Nuno Lopes via llvm-dev wrote: > This program stores 8 bits, and leaves the remaining 24 bits > uninitialized. It then loads 16 bits, half initialized to %v, half > uninitialized. SROA transforms the above function to: > > define i16 @g(i8 %in) { > %v = add nsw i8 127, %in > %1 = zext i8 %v to i16 > %2 = shl i16 %1, 8 > %3 = and

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

On 03/03/2018 12:59 AM, Ilia Mirkin wrote: > On Fri, Mar 2, 2018 at 6:46 PM, Mario Kleiner > <mario.kleiner.de at gmail.com> wrote: >> On 03/02/2018 11:29 PM, Ilia Mirkin wrote: >>> OK, so even if you're passing 1024 to xf86HandleColormaps, gamma_set >>> still only gets called with a 256-entry LUT? If so, that works nicely >>> here, but is not

From: Jyri Sarha <jsarha at ti.com> I optimized Speex resampler for NEON capable ARM CPUs. The first patch should speed up resampling on any platform that can spare the increased memory usage. It would be nice to have these merged to the master branch. Please let me know if there is anything I can do to help the the merge. The patches have been rebased on top of master branch in

On Sat, 2004-09-11 at 12:49, Jeff Cohen wrote: > For the heck of it I tried upgrading to gcc 3.4.2 (from 3.3.3). It > didn't make a difference. So here are the failures for llvm-test. All > diffs are against the "native" output. > > ===================== MultiSource/Applications/sgefa > > cbe failed differently from jit/llc. First cbe: > > 84c84

We have recorded wav files with 44k, 22k, 16k, 11k and 8k Asterisk does not accept these wav files. I used sox input.wav output.gsm to get them to work. However, the only the 8k file did convert and the quality is poor. How can I improve the quality? bye Ronald