similar to: How to enable OPUS inband FEC

Hi Is it possible to cross-compile opus-tools with mingw and Ubuntu? So far I have done this:- # prepare $ mkdir $HOME/source $ mkdir $HOME/builds $ export PATH="$PATH:$HOME/mingw-w64-i686/bin" $ PKG_CONFIG_PATH="$HOME/builds/lib/pkgconfig" # Install ogg $ cd $HOME/source $ svn co http://svn.xiph.org/trunk/ogg $ cd ogg $ ./autogen.sh && ./configure

Hi I have 2 sets of data that I want to do a Wilcoxon test on. They are of the same dimension. One has 4 zero values and the other has 5. > dim(SampA) [1] 1 10 > dim(SampV) [1] 1 10 I get the folowing error Error in wilcox.test.default(SampA, SampV, na.rm = TRUE, paired = FALSE, : 'x' must be numeric I am using the function wilcox.test(SampA, SampV, na.rm=TRUE,

Here's how I set up the data for processing: // For moving data into 32 bit shape uint8_t *buffer8 = NULL; uint16_t *buffer16 = NULL; uint32_t *buffer32 = NULL; unsigned sample32; unsigned sample, channel; uint32_t bitsPerSample = this->get_bits_per_sample(); numFrames = inData.GetSize();

I do the following to init flac: encoder = FLAC__stream_encoder_new(); e = FLAC__stream_encoder_set_do_mid_side_stereo (encoder, numChannels == 2); e = FLAC__stream_encoder_set_loose_mid_side_stereo (encoder, numChannels == 2); e = FLAC__stream_encoder_set_channels (encoder, numChannels); e = FLAC__stream_encoder_set_bits_per_sample (encoder, jmin (24,

--- Roland Rabien <Roland.Rabien@mackie.com> wrote: > I do the following to init flac: > > encoder = FLAC__stream_encoder_new(); > > e = FLAC__stream_encoder_set_do_mid_side_stereo (encoder, > numChannels == 2); > e = FLAC__stream_encoder_set_loose_mid_side_stereo (encoder, > numChannels == 2); > e =

I'm using libFLACC++ and libFLAC and I think that I'm using the calls in the typical order (see code below). But every monoe or stereo file that I send thru I get files that are the same sze as the orginal wave files. Doing a flac -a on the flac files I see that I get: frame=9 blocksize=4608 sample_rate=8000 channels=1 channel_assignment=INDEPENDENT subframe=0

Hi I have managed to do a paired t-test with a data set i have 5 colums of data im dealing with GENE              SampA              SampB              SampC                   SampVehicle ctcc                   859                     na                     145                           24 gtcg                   45                         5                       54                           69

hi all, i'm new to speex and i'm stuck on a problem. I have a buffer that holds 5 seconds of audio data, sampled at 16000 hz and quantitized with 16 bits pcm. I only want to encode it with speex and then decode it and write it back in the same buffer. no compiling errors and no errors during runtime, but from the sample of 5 seconds only a short noise is all I get. I hope you have any

looks fine, I would suspect how the PCM sample are formatted and sent to process(), could you show that part of the code? Josh --- James Smith <jsmith@landmarkdigital.com> wrote: > > I'm using libFLACC++ and libFLAC and I think that I'm using the calls > in the > typical order (see code below). But every monoe or stereo file that > I send > thru I get files

Hi All, I m having a strange problem with the Ogg-Vorbis writting code. The code I m using to write is skipping some samples at the end of the file. For example I m converting the 10000 sample .wav file ( 441000 sample rate , 16 bit depth , stereo ) to ogg format. But while reading the ogg file I only find 5824 samples in the ogg file. Can any one suggest what could be wrong in the code. Is

Hi All, I m having a strange problem with the Ogg-Vorbis writting code. The code I m using to write is skipping some samples at the end of the file. For example I m converting the 10000 sample .wav file ( 441000 sample rate , 16 bit depth , stereo ) to ogg format. But while reading the ogg file I only find 5824 samples in the ogg file. Can any one suggest what could be wrong in the code. Is

Hello Martin, > how a number of samples with no defined size is translated into 8, 16, or 32 bit values in my buffer Although I do not know in detail about the actual implementation of libvorbis, most of the decoders should internally produce single-precision (32bit) floating point values, or 32bit integers as the result of PCM decoding. They are downcast to PCM samples with 8bit, 16bit, or

Hi, I seem to have found a bug with seeking in opusfile in some situations, though this might be better directed to the ogg list. I?ve included a short description below and a test case C program that describes it more detail (comments) and reproduces the issue when used with a file which contains packets that have been split over a page boundary (less common than I thought). The test program is

Okay, how do I drop a changeset/patchset/tag for you folks from SVN? At this point, I have written three examples of how to use the basics of the ogg streaming and decoding in Tremor. I heartily welcome any suggestions, improvements and corrections that you can point out in the code. The examples required me to make some small modifications to the main tremor library. However, the changes

Hello, Is FEC supposed to work in NARROWBAND mode ?(with maxaveragebitrate=12000; maxplaybackrate=8000 ) ?I am having some confusing results, it appears that FEC is enabled in the encoder, but the decoder cannot find any packet with FEC. I am also wondering if this piece of code is correct (webrtc): /* The following is to parse the LBRR flags. */? if (opus_packet_parse(payload,

At this bitrate the encoder likely decides that it's better to put all the bits in the normal packet than use FEC. When you enable FEC it steals a lot of bits from the non-FEC content. Also, the use of FEC depends on the reported percentage of packet loss. The more loss there is, the lower the threshold for enabling FEC. Overall, the encoder attempt to make the best decision on a

I am deeply sorry about the corrupt message just being sent; there seems to have been a compatibility issue with my mailer and my browser. This is an identical copy of the previous message: Hello Martin, Vorbis encoders are lossy, which is in a sense equivalent to converting the sample size of the raw PCM stream into something that would result in the desired bitrate. The "sample size"

flight 17325 linux-linus real [real] http://www.chiark.greenend.org.uk/~xensrcts/logs/17325/ Regressions :-( Tests which did not succeed and are blocking, including tests which could not be run: test-amd64-amd64-xl 9 guest-start fail REGR. vs. 12557 test-amd64-i386-xl-credit2 15 guest-stop fail REGR. vs. 12557 test-amd64-i386-xl 15 guest-stop

flight 17454 linux-linus real [real] http://www.chiark.greenend.org.uk/~xensrcts/logs/17454/ Regressions :-( Tests which did not succeed and are blocking, including tests which could not be run: test-amd64-i386-pair 17 guest-migrate/src_host/dst_host fail REGR. vs. 12557 Regressions which are regarded as allowable (not blocking): test-amd64-i386-qemut-rhel6hvm-amd 7 redhat-install fail

flight 17612 linux-linus real [real] http://www.chiark.greenend.org.uk/~xensrcts/logs/17612/ Regressions :-( Tests which did not succeed and are blocking, including tests which could not be run: test-amd64-i386-pair 17 guest-migrate/src_host/dst_host fail REGR. vs. 12557 Regressions which are regarded as allowable (not blocking): test-amd64-i386-rhel6hvm-intel 9 guest-start.2 fail