similar to: Uncoupled mode?

Hi! I would like to share with you an interesting test result. I re-encoded a song 10 times (the 2. made from the 1., the 3. made from the 2.,... the 10. made from the 9.), with a beta4 (CVS 20010620) and with an RC2 (CVS 20010817) library version, 256 kbit/s mode, channel coupling disabled at RC2. The beta4 is very good at the 10. encoding too, but the RC2 has some interesting quality bugs. So,

After doing an informal (128k) listening test, I have concluded that I prefer Beta4 over RC2. The 16kHz low-pass on the RC2 encoder makes it sound like FM radio. Both encoders SEEM to have a couple of dB bump at 10kHz. JT --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list, send a message to

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

Hello! 1. You should modify the 'vorbis_ftoi(f)' for the undefinied OS/compiler in the 'os.h' on this way: old: static int vorbis_ftoi(double f){ return (int)(f+.5); } new: #define vorbis_ftoi(f) ((int)(f+.5f)) Because in some cases we don't use this function, and the compiler writes many error messages. 2. Could you write a frequency and bitrate independent

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

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

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

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

Hi, I encoded some songs into 350 kbit/s with beta2 and beta3 too, and I don't know why, but it seems the b2 sounds better (makes a better quality ogg file)... The b2 contains more the b3 contains less high sounds. And with the b3, the size of file is smaller with 4-5%... Do you know this problem, or do you think so, that the sound quality is good (or better in the b3) ?? Please be carefull

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:

Hello! Some minor bugfixes... source: psy.c function: _vp_psy_init 1. '-' or '+' in this line? maxoc=toOC((n*.5f - .25f)*rate/n)*(1<<(p->shiftoc+1))+.5f; maxoc=toOC((n*.5f + .25f)*rate/n)*(1<<(p->shiftoc+1))+.5f; 2. I know: sizeof(float *) and sizeof(float) are 4 bytes long, but... old lines: p->noisemedian=_ogg_malloc(n*sizeof(float *));

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

Hello, You know, I always worry about the precision and the float constants... and there is a large lookup table in the floor1.c ... and I figure out a way to keep the code size and speed, but to improve the precision at this lookup table. (the difference is small, but audible) Here is the modifications in the floor1.c: tatic unsigned long FLOOR_fromdB_LOOKUP[256]={

Hello, I've read http://xiph.org/ogg/vorbis/doc/stereo.html and the Vorbis specs, but I haven't got a clear concept of how does the Square Polar Mapping works. I mean, in the "Ogg Vorbbis stereo-specific channel coupling discussion" document, it's described how the decoupling process works for Square Polar Mapping (and it is fairly clear explained), but it does not describe

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

Hi! >From http://www.vorbis.com/stereo.psp : "Oggenc's default choice varies by bitrate and each mode is selectable by the user" So, how am I supposed to do that? E.g. what do I have to do when I want an 128kbps .OGG using lossless stereo, or does this "selectable by the user" mean only that I can choose the stereo mode indirectly by choosing an appropriate bitrate?

Hi; In the Vorbis 1.1 release notes from July 7, 2004, Monty says "The 1.1 libvorbisenc allows setting the fixed reservoir size (in bits, defaulting to two seconds worth of requested bitrate) and 'hoarding' behavior (whether the encoder tends to keep the bit reservoir more full or more empty) as well as the other encoding heuristics available through the API of 1.0.1."

>Yesterday I have finished writing the ambisonic pan filter for oggenc. May I ask what this "pan filter" is? I made some tentative suggestions for coupling Ambisonic B-format in a post "Vorbis Ambisonic coupling" on 4feb07 I gather from the last monthly meeting, that some of you, including Monty, had problems with the phase behaviour of B-format. Would anyone like a

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