similar to: Sampling rate using opusenc

Hello, I want information about the behavior of the --speech option in the opusenc program in opus tools package. The documentation only tells that it optimizes for speech, but what does this mean in terms of sampling frequency, bitrate, etc.? Thank you very much. Nicanor Garcia -------------- next part -------------- An HTML attachment was scrubbed... URL:

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.

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

Hello, I understand it would be better to post several messages with separate topics but I hope I don't cause too much mess if I put it all in a single message this time. To be clear, recently I've been testing Opus Tools under Windows and these are my questions/observations. ????#1. To test encoding at low bitrates, I encoded a sine sweep at 12 kbps with Opusenc and then decoded

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

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

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

--- vorbiscomment/vorbiscomment.1 | 4 +--- 1 files changed, 1 insertions(+), 3 deletions(-) diff --git a/vorbiscomment/vorbiscomment.1 b/vorbiscomment/vorbiscomment.1 index 0108e78..2bceb83 100644 --- a/vorbiscomment/vorbiscomment.1 +++ b/vorbiscomment/vorbiscomment.1 @@ -39,13 +39,11 @@ Reads, modifies, and appends Ogg Vorbis audio file metadata tags. .IP "-a, --append" Append

That's pretty bad. Both DirectSoundCapture and WinMM are capable of recording at 16kHz. I don't know why OpenAL would be incapable of handling it. It's not like it's at all rare or new. I would try 16000 and see if it works. Maybe the docs are wrong? Note that one option to retain high quality is to capture at a higher rate and then downsample using a resampling

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,

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

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

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

> 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)

Hello, oggdec is a very simple ogg decoder, and it fails in the following case: a source A has a fallback A' transcoder curl | oggdec | ices2 is transcoding to lower bitrates ogg A fails, all the listeners are transfered to A' oggdec fails because the stream content changed: [2004-01-29 18:44:47] OggDec 1.0.1 [2004-01-29 18:44:47] Warning: hole in data <p><p>[2004-01-29

It seems that oggdec in Fedora 7 produces a wav header followed by lots of zeros. Building vorbis-tools 1.0.1 from source produces a working oggdec, so I think the problem is in the vorbis-tools package. I can't build vorbis-tools from svn due to some make problems, but I can build the F7 SRPM. There are no differences between the share and oggdec directories, but there are autoconf related

On Sun, Feb 01, 2004 at 07:20:40PM +0100, root wrote: > I have more info about that problem... > curl | oggdec | ices will never work to my opinion with fallback setup on source > because when transfering listeners to fallback, no ogg headers are sent, so > whatever we do, oggdec will not be happy with the data received... I took a look at this. It is true that oggdec does not

Hello there. I can't use bugtracker because it is permanently think that I'm spammer. -------------- System: Windows XP Service Pack 3 When I trying to decode any *.ogg file with oggdec.exe like this: oggdec.exe file.ogg I have an error (it is translate version - I don't have english version of Windows): Instruction from address 0x7c91b1fa trying to access 0x00000010. Memory

Hello, In one of my recent assignments, I was asked to develop a ogg player. I am not able to find the right repository of the source code. There are few repositories on vorbis but I am not clear which one is the right one for Windows environment. If somebody has successfully compiled any ogg player ever, his ideas will be helpful to me. In one of the code set, I was able to compile the

Hello All, I have several soundfiles that are recorded ub 44100Hz, 16-bit Mono. What is the best way and right tools to use to downsample these to 8000Hz so that they can be used with Asterisk. I've tried using sox with the -r switch and Audacity on the mac and Goldwave on Windows and they all generate files that sound like a bad acid trip. I tried increasing the speed 551.25 percent after