similar to: Meaning of mapping[]

On 2014-01-06 10:24 AM, Brendan Bolles wrote: > http://github.com/fnordware/AdobeOgg Cool! > What do the numbers in mapping mean? These numbers tell you how to map the decoded opus channels to particular speakers. > For example, opus_multistream_surround_encoder_create() sends out mapping[] = {0, 4, 1, 2, 3, 5}. > For 6 channels the Vorbis order is front left, center, front right,

I apologize for the terribly long message, but here goes. ------------------------------------------------------------------------------------------------- First, regarding existing tools that I now about. libavcodec and users (e.g. HandBrake): - if there are 6 channels or less, the layout is set by the decoder as per the FLAC specification - if there are more than 6 channels, the layout is

I've been doing a fair amount of work with Vorbis support for Ambisonics, which seems to be going along nicely. It seems that there is signifant interest in coding 5.1 material with Vorbis esp as tarkin becomes more complete, so I've decided to take a break from pure ambisonic work to look into this. I've decided that the best (from a pure elegance and patent avoidance) way to handle

I've been doing a fair amount of work with Vorbis support for Ambisonics, which seems to be going along nicely. It seems that there is signifant interest in coding 5.1 material with Vorbis esp as tarkin becomes more complete, so I've decided to take a break from pure ambisonic work to look into this. I've decided that the best (from a pure elegance and patent avoidance) way to handle

Hello Everyone, I am using the opus 1.1 multistream APIs to encode a 5.1 surround stream on the server, stream it to client, decode it and capture the pcm data. I noticed that there was severe corruption/attenuation on one of the channels(specifically Back/Rear Right). This would appear to be the last channel in the stream. I am attaching an image of the PCM dumps from the original and the one

On Fri Sep 21 16:31:21 2012, Stephen F. Booth wrote: > 6.1: L R C LFE Ls Rs Cs (MPEG 6.1 A layout) I'm confused. WAV puts the rear centre before everything but 'Front left of center' and 'Front right of center'. Are you saying you prefer the extra front channels to to side/rear surround, or that you prefer the MPEG order to the WAVE order? -r

Hey everyone, I've written a FLAC plug-ing for Adobe Premiere. It's actually part of the Ogg Vorbis plug-in I originally set out to write. You can see it here on GitHub: https://github.com/fnordware/AdobeOgg I just finished the first beta a few hours ago. You can download binary plug-ins from here: http://www.fnordware.com/downloads/Ogg_v0.5b1_mac.zip

On 06/01/14 06:21 PM, Ralph Giles wrote: > It's written from the point of view of the decoder, so it seems > backwards if you're thinking about encoding. It may help to think of it > as opus_multistream_surround_encouder_create() returning what you have > to pass to the decoder to understand the stream configuration. Not only that, but it's designed so that the same signal

On Jan 6, 2014, at 11:54 AM, Ralph Giles wrote: > The first entry, mapping[0], > describes front left and contains the value '0', so that speaker is the > left channel of the first coupled stream. mapping[1] is '4', so the > front centre speaker is given the first mono stream. mapping[2] is '1' > so the front right speaker gets the right channel of the first

I get reliable crashes if I create a 6-channel encoder using opus_multistream_surround_encoder_create(). If I use opus_multistream_encoder_create() instead and pass in the same parameters that opus_multistream_surround_encoder_create() sends out (streams = 4, coupled_streams = 2, mapping = {0, 4, 1, 2, 3, 5}) I don't get the crashes. I notice that opus_demo.c uses

Josh Coalson wrote: > --- smoerk <smoerk@gmx.de> wrote: > >>i didn't find anything about tagging flac files as surround files. i >>think there should some possibility to tell the player how to play a >>multi-channel file (how to map the different channels to the speakers >>or if there some decode is needed, like for ambisonic files). > > > it

Hi. I think that vorbis-tools use a wrong order for channels when dealing with 5.1 surround WAV files. More precisely, I find in the Vorbis specification that the output order is: 0: front left 1: front center 2: front right 3: rear left 4: rear right 5: LFE (source: http://www.xiph.org/vorbis/doc/Vorbis_I_spec.html#id342556 ) At the same time, the order of channels in WAV files seems to be:

The FLAC format specification never defined the semantics of 7- and 8-channel files, which has caused some pain for some years now. Attached is a patch to define them. I don't know if this follows "follows SMPTE/ITU-R recommendations," but it follows common tool practice. I chose the set of surround speaker designations used by home theatre systems, which is the same set used by the

I know there's a bug tracker, and that is exactly the reason why I am posting this on the dev list. There has been a report from one of the foobar developers regarding oggenc on the HA forums, but he later posted it at the bug tracker as well. Three weeks have passed and no one seems to have noticed it was there. ~*~ URL: http://trac.xiph.org/cgi-bin/trac.cgi/ticket/736 Subject:

Dear All, I am trying to use the felm function in the lfe package. However it does not seem to deal with missing values the way the lm function does. I wish to tell it na.omit or na.action = na.omit but it does not recognize this. I need to allow for missing values as I have different specifications and don't want to remove observations for all. Help on this will be greatly appreciated!

On 20030610: Gregory Maxwell wrote: (in reply to Ralph Giles) >> I assume you're aware of the technical documentation on dolby's site? >> (http://www.dolby.com/pro/) In particular the surround mixing guide has >> a lot of detailed guidelines. I don't have any practical experience >> with it though, so I can't vouch for it. >Yes I am. I've done a

Note to admins: Please, read through my whole post this time before deleting it. You will be surprised to find out that I have already tried all solutions detailed in the stickies. If you still think that this topic should be deleted, please be kind enough to notify me why, so that I would know how to proceed about getting help with my problem. Thank you. I have 64-bit OpenSUSE 12.1 and I have

Hi All, Are multi-channel (more than 2) formats fully supported in the OggVorbis specification ? I couldn't find any information about multi-channel support on xiph.org. I've used 'oggdropXPd' to encode a 5.1 wavefile and the Xiph OggVorbis libraries (vorbisfile.dll) to decode the file successfully, however the order of the channel interleaving is different to the original wave

Hi , I am trying to optimize OPUS codec to TI compiler C64XX in CCS. I have created the make file and compiled the code in optlevel = 3. The MIPS are pretty high(99 MIPS for 8Khz stereo) for LFE channels and the most MIPS intensive function is the silk_noise_shape_quantizer_del_dec() function. 1. I am trying to convert this entire function to assembly. I have tried to incoporate .asm file but I

Hello, I am trying to connect a instance of ices2 to icecast2. It seems that the connection is done well but I do not know how to take the microphone input to send it to the server. Any help is highly appreciated. Manuel The information is this: Part of my ices.xml: <input> <module>alsa</module> <param name="rate">44100</param> <param