search for: downmixing

This patch adds a new option "channel-mapping" to opusenc which sets the channel mapping family used by the multistream encoder. Please let me know whether adding this option is worthwhile and whether the help string is okay. I tried to keep it short but accurate. The error message for an unimplemented channel mapping is "Error cannot create encoder: request not implemented".

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 extended to have Shelf Filters & Distance Compensation for more sophisticated

Hi! I haven't looked into the code yet, but the patch uses different coding conventions like "if(" and "if ("; like wise "){" and ") {". My personal taste is to have spaces after keywords, but that's just me. I'd prefer a consistent coding style. Regards, Ulrich >>> Michael Graczyk <mgraczyk at google.com> schrieb am 26.04.2016

While integrating 1.1, I ran into some errors 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

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 Sun, 2003-11-23 at 00:35, Kerry Cox wrote: > Now I tried to add a third stream by adding the following <instance> to > the above config: > > <instance> > <hostname>xx.xxx.x.xxx</hostname> > <port>8000</port> > <password>secretpasswd</password> >

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

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 ambisonic...

Folks: I'm trying to set up a streaming server under Fedora Core 2 using Icecast 2.2.0. Everything compiles; everything runs. There are no indications of a problem in any of the log files. When I look at http://192.168.0.128:8000/status.xsl, everything looks ok. But when I click on "Click to listen", it brings up xmms, but there's no indication of audio, and xmms shows a bit

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

...or the remsaple/downmix section below. --> <encode> <quality>0</quality> <samplerate>22050</samplerate> <channels>1</channels> </encode> <!-- stereo->mono downmixing, enabled by setting this to 1 --> <downmix>1</downmix> <!-- resampling. Set to the frequency (in Hz) you wish to resample to, -- > <resample> <in-rate>44100</in-rate> <ou...

...<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 icecast...

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

...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 sure if the current formatting is optimal. I'm referring to (and using terminology from) the struct defined in sect...

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,

...INFO encode/encode_initialise Encoder initialising with VBR: 1 channel(s), 22050 Hz, quality -1.000000 [2002-10-09 01:00:24] INFO audio/resample_initialise Initialised resampler for 1 channels, from 44100 Hz to 22050 Hz [2002-10-09 01:00:24] INFO audio/downmix_initialise Enabling stereo->mono downmixing [2002-10-09 01:00:24] DBUG encode/encode_clear Clearing encoder engine [2002-10-09 01:00:24] DBUG reencode/reencode_page Reinitialising reencoder for new logical stream [2002-10-09 01:00:24] INFO encode/encode_initialise Encoder initialising with VBR: 2 channel(s), 44100 Hz, quality 0.000000 Sh...

...r the remsaple/downmix section below. --> <encode> <quality>0</quality> <samplerate>22050</samplerate> <channels>2</channels> </encode> <!-- stereo->mono downmixing, enabled by setting this to 1 --> <downmix>0</downmix> <!-- resampling. Set to the frequency (in Hz) you wish to resample to, --> <resample> <in-rate>44100</in-rate&gt...

Stack: (gdb) bt #0 0x0000000000aaf38a in silk_VAD_GetNoiseLevels (pX=pX at entry=0x7f26740297a0, psSilk_VAD=psSilk_VAD at entry=0x15897c38) at silk/VAD.c:315 #1 0x0000000000aa4a9d in silk_VAD_GetSA_Q8_sse4_1 (psEncC=0x15897c18, pIn=<optimized out>) at silk/x86/VAD_sse.c:177 #2 0x0000000000a9f92b in silk_encode_do_VAD_FLP (psEnc=psEnc at entry=0x15897c18) at