search for: bandpass

Hi, does anybody have any ideas regarding the easiest and most efficient way of implementing a bandpass filter in R ? any help would be truly appreciated. cheers, Pat Johnston -.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.- r-help mailing list -- Read http://www.ci.tuwien.ac.at/~hornik/R/R-FAQ.html Send "info", "help", or "[un]subscribe&quot...

I have an audio device whose 'driver' gives me Opus encoded frames using OPUS_APPLICATION_AUDIO and max bandwidth set to OPUS_BANDWIDTH_NARROWBAND. How does Opus encoder decide the center point of the 4K bandpass filter? Is it done frame by frame? -- Tony -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/opus/attachments/20150224/a72a3fde/attachment.htm

...e fly. Has anybody on the list tried this? Kelvin Chua On Wed, Mar 4, 2015 at 5:53 PM, Dragos Oancea <droancea at yahoo.com> wrote: > Hi Kelvin, > > You can use something like : > opus_encoder_ctl(enc,OPUS_SET_BITRATE(bitrate)); > opus_encoder_ctl(enc,OPUS_SET_MAX_BANDWIDTH(bandpass)); > bandpass is the audio bandpass , eg: OPUS_BANDWIDTH_WIDEBAND . > > You will need to calculate the codec bitrate from the available network > bitrate (by taking into account the size of the IP+UDP+RTP headers which is > 40 bytes ) . > By default the audio bandwidth (bandpass)...

...s take a subset of the vector produced by the FFT and go from that back to the initial data frame of [timestamp,events], from which I can go back to the source data and pull out the IPs and so forth. How can I do this? My roommate (an aero engineer) has recommended that I look into implementing a bandpass filter but I'm finding the material to be way over my head. I'm at the limits of my expertise (I am not an engineer :) so if anyone just wants to point me at the topics I need to study, that would be fine--although pointing out some specific examples would be great as well. Thanks,...

I am using libopus for my implementation. I wonder if anybody in the list have any experience on how to make libopus dynamically adjust its bitrate? On Mar 3, 2015 10:42 PM, "Benjamin Schwartz" <benjamin.m.schwartz at gmail.com> wrote: > It sounds like your software isn't adjusting the opus bitrate in response > to network conditions. For example, many WebRTC

Hi Kelvin, The audio bandpass setting is only configurable when the encoder is instantiated (eg: start of a Voip call ) , but you can change the bitrate anytime.For example if you can read incoming RTCP packets , you can check if there's reported packet loss , and then lower the bitrate accordingly.Yes, the app has to be aw...

Dr. Williams, I ran across your inquiry on one of the R-help mailing lists regarding digital filter design and implementation. I found no response to your email in the archives and was wondering if you were able to find anything. Thanks, Israel -- Israel Christie, Ph.D. Email: ichristie at gmail.com Phone: 865.766.0214 Mobile: 865.406.4615

Hello list, Is there any way to bandpass filter in R thanks nuncio -- Nuncio.M Research Scientist National Center for Antarctic and Ocean research Head land Sada Vasco da Gamma Goa-403804 [[alternative HTML version deleted]]

Hi Kelvin, You can use something like :opus_encoder_ctl(enc,OPUS_SET_BITRATE(bitrate));opus_encoder_ctl(enc,OPUS_SET_MAX_BANDWIDTH(bandpass)); bandpass is the audio bandpass?, eg: OPUS_BANDWIDTH_WIDEBAND . You will need to calculate the codec bitrate from the available network bitrate (by taking into account the size of the IP+UDP+RTP headers which is 40 bytes ) . By default the audio bandwidth (bandpass) setting is OPUS_BANDWIDTH_FULL...

> > I am talking about the original sweep. > > The original sweep stops pretty close to 24 kHz. I mean the original sweep _as_encoded_, sorry.

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

I have a time series of data from an electroencephalogram (EEG). I wish to filter the data to get rid of 50Hz mains 'hum'. I have 'designed' a combination bandpass and notch filter using a web- site. The site returns the filter in "ANSI C" source code. It is:- /* Digital filter designed by mkfilter/mkshape/gencode A.J. Fisher Command line: /www/usr/fisher/helpers/mkfilter -Bu -Bp -o 5 -a 1.0000000000e-02 1.0000000000e-01 -Z 2.0000000000e-01 -l...

Hey, I am having trouble with the size of the encoded bytes by Opus. I am also having issue with the Bandwidth ctl. Here is the scenario. If I encode 16khz sampled audio: opus_encoder_ctl(enc,OPUS_SET_MAX_BANDWIDTH(OPUS_BANDWIDTH_WIDEBAND)) opus_encoder_ctl(enc, OPUS_GET_BANDWIDTH(&x)) = 1102 opus_encoder_ctl(enc, OPUS_GET_MAX_BANDWIDTH(&x)) = 1103 average encoded size = 120 bytes if I

...for long/short blocks ? Why are there so many possible blocksizes - would it be necessary to analyze the entire track to find the out, which sizes fits best ? 2. How does the encoder decide , whether to use short or long blocks during encoding ? In earlier versions, it was done "by 4 parallel bandpass filters that detect energy surges", but Monty said it will be changed. (http://www.xiph.org/archives/vorbis-dev/200109/0054.html ) Thanks Stoffke <p>--- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this lis...

...ay" command easily at the command line. Also when I type "sox -h" I get: sox: Version 12.17.7 Usage: [ gopts ] [ fopts ] ifile [ fopts ] ofile [ effect [ effopts ] ] gopts: -e -h -p -v volume -V fopts: -r rate -c channels -s/-u/-U/-A/-a/-i/-g/-f -b/-w/-l/-d -x effect: avg band bandpass bandreject chorus compand copy dcshift deemph earwax echo echos fade filter flanger highp highpass lowp lowpass mask mcompand noiseprof noisered pan phaser pick pitch polyphase rate repeat resample reverb reverse silence speed stat stretch swap synth trim vibro vol effopts: depends on effect S...

...the list of supported file formats as "mp3". [root at host0040 ~]# sox -h sox: Version 12.18.1 Usage: [ gopts ] [ fopts ] ifile [ fopts ] ofile [ effect [ effopts ] ] gopts: -e -h -p -q -S -V fopts: -r rate -c channels -s/-u/-U/-A/-a/-i/-g/-f -b/-w/-l/-d -v volume -x effect: avg band bandpass bandreject chorus compand copy dcshift deemph earwax echo echos fade filter flanger highp highpass lowp lowpass mask mcompand noiseprof noisered pan phaser pick pitch polyphase rate repeat resample reverb reverse silence speed stat stretch swap synth trim vibro vol effopts: depends on effect Supp...

..."low". But when I reversed the Wp and Ws parameters, it resulted in an object with an empty $type instead of $type being "high". I looked at the source code and found an obvious reason why: if (length(Wp) == 2) { warning("buttord seems to overdesign bandpass and bandreject filters") if (any(stop)) type = "stop" else type = "pass" } else { ## here we set to "high" or "low", as expected if (any(stop)) type = "high&...

...ata deleting data/1934-638.1344.04nov27/vartable deleting data/1934-638.1344.04nov27/leakage deleting data/1934-638.1344.04nov27/history deleting data/1934-638.1344.04nov27/header deleting data/1934-638.1344.04nov27/gains deleting data/1934-638.1344.04nov27/flags deleting data/1934-638.1344.04nov27/bandpass deleting data/1934-638.1344.04nov28/visdata deleting data/1934-638.1344.04nov28/vartable deleting data/1934-638.1344.04nov28/leakage deleting data/1934-638.1344.04nov28/history deleting data/1934-638.1344.04nov28/header deleting data/1934-638.1344.04nov28/gains deleting data/1934-638.1344.04nov28/f...

I hope it is OK to ask an Opus question here. If not, please advise the correct forum. Regarding the statement "Internally, the codec always operates at a 48 kHz sampling rate, though it allows input and output of various bandwidths..." Can I infer from this that it is most CPU efficient to always input and output PCM data at 48000 samples per sec, regardless of other factors such as

...enter point"? Jean-Marc On 24/02/15 11:30 AM, Tony wrote: > I have an audio device whose 'driver' gives me Opus encoded frames > using OPUS_APPLICATION_AUDIO and max bandwidth set to > OPUS_BANDWIDTH_NARROWBAND. How does Opus encoder decide the center > point of the 4K bandpass filter? Is it done frame by frame? > > -- > Tony > > > _______________________________________________ > opus mailing list > opus at xiph.org > http://lists.xiph.org/mailman/listinfo/opus >