search for: downmix

...fault: 20)\n"); printf(" --expect-loss Percentage packet loss to expect (default: 0)\n"); + printf(" --channel-mapping Channel Mapping Family\n"); + printf(" (default: {0, 1, 255} for {1-2, 3-8, 9+} channels)\n"); printf(" --downmix-mono Downmix to mono\n"); printf(" --downmix-stereo Downmix to stereo (if >2 channels)\n"); printf(" --max-delay n Maximum container delay in milliseconds\n"); @@ -260,6 +262,7 @@ int main(int argc, char **argv) {"complexity", required_arg...

A strategy for downmix support a) simple 4.0, 5.0, 6.0 7.0 Speaker Decode on the player via a separate library from the Vorbis core. This is the easy panning solution on the Wiki Ambisonic Decoder page and requires hardly any computing power. Most important cos its all about surround init? This library can be extende...

...rintf(" --expect-loss Percentage packet loss to expect > (default: 0)\n"); > + printf(" --channel-mapping Channel Mapping Family\n"); > + printf(" (default: {0, 1, 255} for {1-2, 3-8, > 9+} channels)\n"); > printf(" --downmix-mono Downmix to mono\n"); > printf(" --downmix-stereo Downmix to stereo (if >2 channels)\n"); > printf(" --max-delay n Maximum container delay in milliseconds\n"); > @@ -260,6 +262,7 @@ int main(int argc, char **argv) > {"complexity...

...with my compiler and, after examining the code, saw what might be some invalid pointer math in opus_encoder.c. In optimize_framesize(), the first parameter 'x' is typed as opus_val16 and pointer math is done on the variable (in the 'if (buffering)' block) and then x is passed to the downmix function. I see that the downmix function uses the proper type, but it appears to me that the pointer adjustment performed in that 'if (buffering)' block is always using 16-bit. Kevin O'Connor

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

On Aug 07 08:30:31, hans at stare.cz wrote: > On Aug 07 00:41:52, petrparizek2000 at yahoo.com wrote: > > ????#1. To test encoding at low bitrates, I encoded a sine sweep at 12 kbps > > with Opusenc and then decoded the resulting file with Opusdec. > 1) Opusenc --bitrate 12 --downmix-mono Sweep50.wav Sweep50.opus Why are you using a stereo file containing the same sweep in both channels and then downmixing to mono? Also, why are you repeating the sweep twice in every channel? Also also, the file you linked to is Sweep50FlatMidSide.wav as opposed to Sweep50.wav - is it really...

...t>/smallband.ogg</mount> > <yp>0</yp> > <reconnectdelay>2</reconnectdelay> > <reconnectattempts>5</reconnectattempts> > <maxqueuelength>80</maxqueuelength> > > <downmix>1</downmix> > > <resample> > <in-rate>22050</in-rate> > <out-rate>11000</out-rate> > </resample> > > <encode> > <nominal-bitrate>160...

The prospect of people actually putting B-format audio (via the panner or directly input) into Ogg/Vorbis brings an interesting challenge: What do we do with the audio after decoding it? The following sane options exist: A) Simply output the B-format audio B) Produce a downmix 1) Mono. 2) Stereo blumlein crossed pairs 3) Stereo UHJ 4) binaural C) Produce speaker feeds 1) Fully generalizable speaker feed decoder (such as http://www.kokkinizita.net/linuxaudio/adec-pict.html) 2) G-format (fixed decode for the 5.1 layout) (A) is pretty much a no-brainer,...

First let me say that I was absolutely astounded at the sound quality when using oggenc at q = -1 (around 50 kbps). So much so, that I'm anxious to try some of the lower bit rates that were mentioned in the announcement for 1.0: ... audio and music at fixed and variable bitrates from 16 to 128 kbps/channel. But I am having trouble finding how to use either oggdrop or oggenc to get these

...trivial. Usually, decoding ambisonics is the job of a dedicated > decoder, and depends on the role, number and positions of output > channels. so I don't think that the Opus decoder should be involved, > unless approximate methods could be considered good enough as default. I included downmixing to stereo because it is simple and there is a clear right way to do it. For 5.1 or other surround setups, there is no obvious best solution so I did not include those downmixing matrices. The purpose of this downmixing matrix is to give guidelines for what to do in the likely case that an ambiso...

...assword> <mount>/midband.ogg</mount> <yp>0</yp> <reconnectdelay>2</reconnectdelay> <reconnectattempts>5</reconnectattempts> <maxqueuelength>80</maxqueuelength> <downmix>1</downmix> <resample> <in-rate>44100</in-rate> <out-rate>22050</out-rate> </resample> <encode> <nominal-bitrate>32000</nominal-bitrate>...

Can someone who really knows the Ogg command-line encoder, help recommend the best setting for 33.6k modem stereo music streaming? (56k doesn't count cuz many people's 56k modems don't work at a full 56k, and I want them to be able to surf CD Baby at the same time as listening. 2 minutes / 120 seconds of audio should be about 400k.) I'm at my wit's end: tried everything I

...;!-- allow stream to be advertised on YP, default 0 --> <!-- Live encoding/reencoding: channels and samplerate currently MUST match the channels and samplerate given in the parameters to the oss input module above or the remsaple/downmix section below. --> <encode> <quality>0</quality> <samplerate>22050</samplerate> <channels>1</channels> </encode> <!-- stereo->mono downmixing, enabled by s...

...<port>9000</port> <password>********</password> <mount>/egoplay24.ogg</mount> <yp>1</yp> <!-- allow stream to be advertised on YP, default 0 --> <!-- stereo->mono downmixing, enabled by setting this to 1 --> <downmix>1</downmix> <resample> <in-rate>44100</in-rate> <out-rate>22050</out-rate> </resample> <encode>...

On 2006-06-21T15:21+0100 Karl Heyes wrote: > You capture at 48000 but resample from 44100, that will cause problems. > You resample to 22050 so that will affect quality, you haven't said what > bitrate you want to target for, so resampling/downmixing may be required > anyway. I am unsure about these settings. Any particular suggestions? I don't quality with voice, just clarity. > >Can I cut the delay? > A few secs delay is generally not a problem, but if you want to reduce > it then reduce the burst-size (eg 0) in icec...

Hi, I would like to create an output file with mono audio. How to do it ? If use a video with mono as input, the result will be a video with stereo. Is there a way to use the --downmix from oggenc ? Regards -- ZikZak

...names into the spec, or cite some other document from a standards body naming and numbering speaker positions (does anyone know anything applicable?). Should this list be made into its own registry, for later extensibility? ffmpeg additionally defines "stereo left/right" (for an embedded downmix), "wide left/right", "surround direct left/right", and "low frequency #2". The downmixing algorithm is largely a description of ffmpeg's libswresample's behavior. Documenting some downmixing behavior was suggested by mark4o in the #opus channel on Freenode; not...

darkice is an execellent bit of software, very well written. and the author of the darkice software, akos, is very active on the darkice ML. http://yp.tilos.hu:9000/tilos_high.ogg <= for a darkice stream. b. On Sat, Nov 22, 2003 at 12:05:55PM -0500, Brendan Cully wrote: > On Saturday, 22 November 2003 at 10:50, Kerry Cox wrote: > > So basically, I download and install ices 0.3,

...and the like. I recompiled everything this morning and set it running again. But one mount has died already. We have 3 streams running. the first is the ogg files as they were encoded (egoplay.ogg). the second is a re-encode to 44.1khz stereo at quality 0 (egoplay64.ogg). and the third is a downmix and resample to 22.05khz mono at quality -1 (egoplay24.ogg). It's this latter one that's died. by using the "current song" information from the stats page, I was able to find the log entries around the time it died. Here's what the ices logfile shows: [2002-10-09 01:00:24...

...!-- allow stream to be advertised on YP, default 0 --> <!-- Live encoding/reencoding: channels and samplerate currently MUST match the channels and samplerate given in the parameters to the oss input module above or the remsaple/downmix section below. --> <encode> <quality>0</quality> <samplerate>22050</samplerate> <channels>2</channels> </encode> <!-- stereo->mono downmixing, enabled by...