similar to: Multi-channel support

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

Could I request, please, that the link on this page that currently points to: http://homepage.ntlworld.com/jfe1205/OggVorbis/ be changed to point to: http://www.rarewares.org/ogg-oggdropxpd.php It is simply that the Rarewares site can support a much greater bandwidth than I can and I usually update it ahead of my own webspace. (It's also a 'proper' web site rather than just an ftp

lvqcl wrote: > Martin Leese wrote: >> Note that the channel order may not be defined. > > IMHO it doesn't matter in this place of documentation (which describes > default channel assignments for FLAC). Your proposed wording was: 0000-0111 : (number of independent channels)-1. The channel order follows SMPTE/ITU-R recommendations. The assignments are as follows: The

I released "aoTuV" beta6 of OggVorbis encoder, library. ---------------------------------------- CHANGES (for general) from beta5.7 # Libvorbis aoTuV was unified with Xiph.Org's libvorbis1.3.2. # Pre(post)-echo of the specific pattern was improved. # The bit allotment when sound of the different type coexisted was improved. # The calculation method of dynamic thresholds

lvqcl wrote: ... > From FLAC 1.2.1 documentation: > > 0000-0111 : (number of independent channels)-1. Where defined, the channel > order follows SMPTE/ITU-R recommendations. The assignments are as follows: > 1 channel: mono > 2 channels: left, right > 3 channels: left, right, center > 4 channels: left, right, back left, back right > 5 channels: left, right, center,

Hi i'm developing a sort of VoIP application for my ipaq using speex... I'm still at beginning and i have many problems encoding and decoding my wav files....output is only noise! Why? I'm using Libspeex 1.1.3, Embedded VisualC++ 3.0, Ipaq 3850(206 MHz Intel® Strong ARM 32-bit RISC Processor) PocketPC 2002 (Windows CE 3.0). Libspeex is complied with the definition of

Now that we've selected a channel mapping for 7 and 8 channel flac, the command-line encoder tools needs updating to accept wav files with compatible channel maps. -r -------------- next part --------------

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

I compared quicktime aac and vorbis,i think quicktime aac is better than vorbis at 80Kbps. please tell me if i'm wrong. using command: qtaacenc.exe --tvbr 31 --highest --samplerate keep test.wav qt.m4a oggenc2.exe --raw -q 1.6 test.wav -o vorbis.ogg the version: qtaacenc version 20100725 with QuickTime 7.6.7 OggEnc v2.87 (libvorbis 1.3.1) links:

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

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

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). any idea how to implement it?

I need to encode a WAV to OGG but must resample to 22050 samples per second, but can't do that with oggdropXPd.exe. The application always crashes. What I do is: Start oggdropXPd.exe. Right-click on fish, choosing 'Encoding Options'. Under Advanced Options, I set Re-sample to 22050 samples per second. Click 'Accept'. Drag WAV on fish. In the directory where the WAV originates,

The commit http://git.xiph.org/?p=flac.git;a=commitdiff;h=99257e177eac96fa41a107b423080232f59ebe45 also requires some changes in write_iff_headers() function: currently flac don't write WAVEFORMATEXTENSIBLE header if decoder_session->channel_mask is equal to 0, 1 or 3. After the commit 99257e17 flac should do this for channel_mask equal to 0, 4 or 3. The patch fixes this.

Hi (again), I started to work on an ACM (Audio Compression Manager) for FLAC. I currently only have the canvas. The encoding/decoding is not done yet. So it's completely useless. But since the canvas is the harder thing to start, here it is. (just in case I don't have time to continue) http://mukoli.free.fr/flac-ACM.v0.1.0.zip

Are the ogg/vorbis programs updated to v1.0.1 ? OggDrop for example ? Regards, David ---------------------------------------------------------------------------- ----------- David Balazic mailto:david.balazic@hermes.si HERMES Softlab http://www.hermes-softlab.com Zolajeva 30 Phone: +386 2 450 8851 SI-2000 Maribor Slovenija

On Dec 14, 2014, at 10:02 AM, lvqcl <lvqcl.mail at gmail.com> wrote: > Currently the header of a decoded WAV file can be different to the > original WAV file because FLAC doesn't preserve 'fmt ' chunk. > > For example: create a 24-bit stereo .wav file with WAVEFORMATEXTENSIBLE > header with channel mask == 3. Encode it to .flac then decode back to .wav. > FLAC

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

Hi (hope I am in the right mailing list!) I really can't figure out how to use Vorbisgain in OggdropXPd - it doesn't seem to work so far. Can Someone pls. explain detailed (steps) what to do when using Vorbisgain in OggdropXPd? I understand the settings for Vorbisgain are in "decoding options"... why? What I really want is the function of Mp3Gain on Ogg-files, and I thought