search for: uncoupling

Hello, A question: Can we use uncoupled encoding in the 1.0? (current CVS) How? Because I see some uncoupled definitions (books\uncoupled\res_books_uncoupled.h), but I don't know how could I switch to it manually. Perhaps with an option in the vorbis_encode_ctl() function... Because I think so it's better if I don't use any kind of channel coupling (lossless neither) on high bitrates

I have just started trying Opus with a view to using it in a project. I am interested in embedded hardware and tried it on the Raspberry Pi using the raspbian distro. The version of libopus in the repos is 0.9.14. I installed this and tried encoding 2 minutes of speech from a librevox recording. It managed this at a respectable pace for complexity 10: Skipping chunk of type "LIST",

Christian, I will give 64kbit/s a try and post the figures. My own project is voice only and requires low bitrate so was hoping that it was just the way I was compiling and not an actual regression in speed for SILK. The raspberry PI is quite a cheap and handy reference platform though the ARM side is fairly underpowered but has a great GPU. It also has no audio in which is a pain for playing

Hi Stuart, you are compressing it at 6kbit/s. Then, then SILK mode is probability used and the Silk mode is much faster than CELT. Do you also some figures at 64kbit/s? It is strange that Opus 1.1 got slower in the Silk mode - may the speech/voice selection adds some overhead. I would be interested in seeing the performance of the 64 kbit/s in both Opus 1.0 and Opus 1.1. With best

I had to change the following to get the 'make distcheck' to work and to be able to build the rpms with 'rpm-build -tb *.tar.gz' I also added a test in libvorbis configure.in to detect older ogg versions (check for oggpack_writealign) and print a meaningful error. I also updated the .spec-file with dependencies on libogg-1.0 Thanks for great work everyone! /noa -- begin:vcard

Sorry for the lack of reproducible data, but this seems to be a problem inherent to my dataset and I can't figure out where the issue is. I have several data frames set up as a time series with identical POSIXct date formats. If I keep the original data in data frame format and merge them using base merge- everything is perfect and everyone is happy. If I transform the data frames to zoo

I had expected that the default bitrate for the encoder would be the same as setting it to OPUS_AUTO, but I'm getting difference results: >opusenc --comp 4 sample.wav sample.opus Encoding using libopus 1.3-rc2 (audio) ----------------------------------------------------- Input: 8 kHz, 1 channel Output: 1 channel (1 uncoupled) 20ms packets, 25 kbit/s VBR Preskip: 312

Resampling to 48khz speeds them both up but the disparity is about the same: 2.609 to 3.69. Best Regards, Stuart Marsden On 17 December 2013 17:04, Stuart Marsden <stuartmarsden at finmars.co.uk>wrote: > Christian, > > Complexity 0, 6kbps: > > 0.9.14 Speed 5.204 > 1.1 Speed 5.218 > > A slight win on that run but they vary enough to say about the same. At >

Here is the modified patch. I added a flag to configure.ac which is set to 0 to disable ambisonics, and 1 to enable it. Right now the implementation simply creates a surround encoder with N uncoupled streams. Thanks, Michael Graczyk -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi Opus list. I subscribed because your discussion on the IETF draft ("Ambisonics in an Ogg Opus Container") was mentioned on the sursound list. I tried Opus for ambisonics more than a year ago. It does works with uncoupled channels (I had to patch the encoder). I don't know what else could be done to optimize support for ambisonics, as I'm not a codec expert. So I think that

...*qA = -*qA; } } There are some things which I don't understand. I think I understand the first half where we decide which channel is larger and swapping them if B is larger than A, etc. If swapping B with A was needed in the decoder, how does the decoder know to swap them back when uncoupling? Secondly, I dont quite get how we can uncouple losslessly if that last if statement were true. That is: if (*qB > fabs(*qA) * 1.9999f) { *qB = -fabs(*qA) * 2.f; *qA = -*qA; } I used some test cases of qA and qB but it seems the uncoupling routine in mapping0.c can't ge...

I just recently found that opus_multistream_encode_float is returning -1 (OPUS_BAD_ARG) with the libopus 1.1 build but works just fine with the libopus 1.0.1 and libopus 1.1-beta builds. I tried using opus_multistream_encoder_create and opus_multistream_surround_encoder_create. Tried with coupled and uncoupled quadraphonic and uncoupled stereo encodes. I'm dynamically loading the libopus

This patch adds top level functions to create an ambisonic multistream encoder. The implementation currently just calls the analogous surround sound functions with channel mapping 255 to create an encoder that bundles uncoupled streams. Forthcoming patches will actually set channel bitrate and other configuration. My main concern is that adding additional

Unfortunately, ambisonics aren't exposed in opusenc yet, thus the trouble. They're an API-only feature, but it's a good time to discuss what such a command-line interface would look like, notably: how to specify multiple streams & stream order, select the mapping family, coupled channels, and how to specify the matrix (for family 3). Likewise, there's no multistream support at

On Sat, 28 May 2016 16:21:33 -0700, Michael Graczyk <mgraczyk at google.com> wrote : > Hi Marc, Hi Micheal. > On Sat, May 28, 2016 at 10:44 AM, Marc Lavallée <marc at hacklava.net> > wrote: > > I subscribed because your discussion on the IETF draft ("Ambisonics > > in an Ogg Opus Container") was mentioned on the sursound list. > > Thanks for

The FAQ says (at its very end): > Does Vorbis support mid/side stereo modes? More than stereo? > > Vorbis has hooks for mid/side stereo and continuous balance stereo, > but the channel mappings for both are yet to be formally specified and > implemented. Currently, Vorbis channels are uncoupled. > This should be updated so that people know how good is Vorbis :) > Vorbis does

Hello, I am testing the command line based opus encoder/decoder tools and I have some questions regarding the information given by the encoder (see below as an example). Encoding using libopus 1.0.2 (audio) ----------------------------------------------------- Input: 48kHz 1 channel Output: 1 channel (1 uncoupled) 20ms packets, 64kbit/sec VBR Preskip: 312 Encoding complete

The iommu is enabled by default when xen is booting and later disabled in iommu_setup() when no iommu is present. But under some circumstances iommu code can be called before iommu_setup() is processed. If there is no iommu available xen crashes. This can happen for example when panic(...) is called as introduced with the patch "x86-64: detect processors subject to AMD erratum #121 and

Hi! Playing with Opus 1.3 I converted a tone sweep with a sample rate of 96kHz (just for fun). Before I had converted that from WAV to FLAC, and to Vorbis without problems. With Opus I noticed that the file size for 48kHz and 48 kbps compared to 96kHz Vorbis at 31kbps is about double the size and it sounds even worse (than Vorbis) (there is a lot of noise in the lower frequencies when a low

Hi, I am a Xen newbie and have been trying to get Xen-3.4.1 working on my Ubuntu Lucid (kernel version 2.6.32-21) system, using kernel 2.6.18.8 as dom0. In attempting this, I ran into the following issue. On booting into Xen, I get these errors: Mounting none on /dev failed. No such device. ... Gave up waiting for root device. ... Alert! Could not mount /dev/sda1 as root device. Dropping to a