search for: decorrelation

Please, I'm looking at speex rc1 echo_cancellation API docs and I found there is now a funcion for doing decorrelation: void speex_decorrelate( SpeexDecorrState * st, const spx_int16_t * in, spx_int16_t * out, int strength) How does it fit to the speex_echo_cancellation function ? When removing echo from multiple channels I need to decorrelate the input mic first and then put the decorrelation...

...;m studying music production and am currently doing an analysis of the FLAC format. If anyone has the knowledge and a minute to explain i would greatly appreciate any help. One thing that i cannot make sense from in the FLAC documentation (a thing that is hard to find info on in general) is how the decorrelation fase works exactly. I understand that the two channels in a stereo file can have similarities that can be exploitet when compressing, but why is this transformation used: mid = (left + right) / 2, side = left - right? Is it simply to straighten out the signals and thereby having less information to...

...cho cancellation I have been searching documentation about that but I could not find it Where it is? If it doesn't exist? Could you add a short example showing how do you think it should work? or It's "just" necessary to: init the multichannel echo cancellation state, init the decorrelation state decorrelate the far-end signal apply cancellation (the same way it is applied to a single channel) destroy the decorrelation state destroy the multichannel echo canceller state, Best regards David -------------- next part -------------- An HTML attachment was scrubbed... URL: ht...

...oduction and am currently doing an analysis of > the FLAC format. If anyone has the knowledge and a minute to explain > i would greatly appreciate any help. One thing that i cannot make > sense from in the FLAC documentation (a thing that is hard to find > info on in general) is how the decorrelation fase works exactly. I > understand that the two channels in a stereo file can have > similarities that can be exploitet when compressing, but why is this > transformation used: mid = (left + right) / 2, side = left - right? > Is it simply to straighten out the signals and thereby having...

Dear all, I made a simple test of the Cholesky decomposition in the package 'Matrix', by considering 2 variables 100% correlated. http://blogs.sas.com/content/iml/2012/02/08/use-the-cholesky-transformation-to-correlate-and-uncorrelate-variables/ The full code is below and can be simply copy&paste in the R prompt. After uncorrelation I still have a correlation of +-100%...

when i encode i multi channel file, does flac encode every channel seperatly or does is look for similarities between channels? the documentation says: INTER-CHANNEL DECORRELATION In the case of stereo input, once the data is blocked it is optionally passed through an inter-channel decorrelation stage. The left and right channels are converted to center and side channels through the following transformation: mid = (left + right) / 2, side = left - right. This is a lossle...

On 1/2/07, Josh Coalson <xflac@yahoo.com> wrote: > I'm not sure I understand... I think the INDEPENDENT you are referring > to means the interchannel decorrelation method. this stage is before > and independent of LPC analysis. Correct. > I'm not sure why "independent" frames would not decode correctly > though. more info on exactly how it is not working would help. Ignoring the fact that I am not doing anything special with the MI...

Hello, I am new to the list, to FLAC and to FLAC development but I am having a problem in trying to make FLAC work with MediaMVP based the mvpmc project (ref. http://www.wvpmc.org) that hopefully someone can help me with. This device is possibly minimum spec. for a FLAC decoder. In any case, I have had some success cross-compiling libFLAC (without the ogg or metadata modules).for the embedded

Hey everyone, I have a few questions about FLAC that aren't answered in the FAQ or the documentation. Maybe someone knows about these details. 1. What exactly is meant by "reference encoder"? 2. Concerning interchannel decorrelation, what is the point of creating separate signals, why not just keep the channels you have? For example, you might create "mid" and "side" channels, but you still end up with 2 so what's the point? The only reason I can think of is maybe a couple of bits are saved by storing...

Hi, I work for a company which makes meteor and wind radar (http://www.gsoft.com.au). On occasion (ie during meteor showers such as the Leonids) we configure the system to save raw data as it comes out of the acquisition system, the data rate for this varies (depends on acquisition parameters and number of coherent integrations etc), but usually it is around 600kb/sec. The data consists of 16

Josh Coalson wrote: > yeah, flac doesn't have a 'gzip' fallback method > so any non-audio data will probably get stored > verbatim. I'm kind of reluctant to add a generic > compressor. If you wan't, you could come up with a > FLAC metadata block to store a gzip'ed chunk and I > could add that to the format. > I had the same thought when I was

Hi everyone, I've just released Speex 1.2rc1. This adds support for acoustic echo cancellation with multiple microphones and multiple loudspeakers. It also adds an API to decorrelate loudspeaker signals to improve multi-channel performance. In the bugfix department, there are fixes for a few bugs in the echo canceller, jitter buffer and preprocessor. At this point, the API for 1.2 should be

...all setup, 2 desktop loudspeakers and have very good results (20 to 30dB rejection) with mono music and speex_echo_state_init_mc() set to 1 speaker but nearly no cancellation with a stereo setup. Initializing echo_state with 2 speakers doesn't improve but rather degrade cancellation. Also tried decorrelation but no noticeable effects. And of course sounds are strongly dynamically panned over the 5 loudspeakers. So here are the questions, is there a hope aec can do something for her or do I am dreaming? do I miss something? Surely stupid but as the loudspeakers and the microphone are always at the same...

> > the best thing would be to try and set the > blocksize > > to match the length of the individual 'sample'. > if > > each sample is much shorter than the blocksize > then > > the encoder may not be able to generate an > efficient > > model of the signal. if the samples within the > > soundfont vary greatly in length that also makes >

Hello, I'm new to this list, and I would like to know about the current status of FLAC and surround formats (5.1, 7.1). I know it is possible to compress interleaved multichannel files with FLAC, but does it in any way have any multichannel matrix, that collects common information from all channels and use this when compressing? If not, wouldn't it be a great idea to have this feature!!!

...t on this device) > which would explain how to transform the INDEPENDENT sample frames > to LPC. Obviously preserving lossless is important for this logic too. > > Thanks in advance. I'm not sure I understand... I think the INDEPENDENT you are referring to means the interchannel decorrelation method. this stage is before and independent of LPC analysis. for stereo data, the encoder chooses whether to encoder the L R channels as is, or to compute mid (M) and side (S) channels and send M/S L/S or R/S. after that, each of the 2 channels can be encoded with LPC or other methods. e.g. 1s...

--- MVallevand <mvallevand@gmail.com> wrote: > On 1/2/07, Josh Coalson <xflac@yahoo.com> wrote: > > I'm not sure I understand... I think the INDEPENDENT you are > referring > > to means the interchannel decorrelation method. this stage is > before > > and independent of LPC analysis. > > Correct. > > > I'm not sure why "independent" frames would not decode correctly > > though. more info on exactly how it is not working would help. > > Ignoring the fact tha...

...he case then the highest -l value one can 'portably' use would be 12? FWIW, I've tried 16 & 12 and noticed no difference in exhaustive's execution time. Another "nit" I noticed, from http://flac.sourceforge.net/documentation_format_overview.html in the inter-channel-decorrelation section -- it mentions -m vs. -M, where -m always uses mid+side channel whether it is bigger or not, while -M adaptively uses mid+side or stereo depending on which is better. Why would flac levels 7 and 8 use -m (always when -M allows adaptive encoding based on which is better? ------------...

...explained at: http://flac.sourceforge.net/format.html in the bulleted point beginning, "FLAC specifies a subset of itself as the Subset format." ... > Another "nit" I noticed, from > http://flac.sourceforge.net/documentation_format_overview.html > in the inter-channel-decorrelation section -- it mentions -m vs. > -M, where -m always uses mid+side channel whether it is bigger or not, > while -M adaptively uses mid+side or stereo depending on which is better. > > Why would flac levels 7 and 8 use -m (always when -M allows adaptive > encoding > based on which i...

Hello, We (Google) have been experimenting with configuration and adjustments to CELT-only Opus that give good results for compressing ambisonic audio signals [1]. Based on our results so far, we would like to use Opus to encode spatial audio. We hope to make it easy/possible to use libopus with other common tools and software modules (ffmpeg/libav in particular). Based on my reading of the