similar to: Stereo channel separation

opusdec -V opusdec opus-tools f2a2e88 (using libopus unknown) I've got an opus file encoded from a .wav off a cd, 44100Hz: opusinfo 2-24-Overture_in_C_\(In_Memoriam\).opus Processing file "2-24-Overture_in_C_(In_Memoriam).opus"... New logical stream (#1, serial: 38134f1f): type opus Encoded with libopus unknown User comments section follows... ENCODER=opusenc from opus-tools

Hi! Playing with Opus 1.3 I converted a tone sweep with a sample rate of 96kHz (just for fun). Before I had converted that from WAV to FLAC, and to Vorbis without problems. With Opus I noticed that the file size for 48kHz and 48 kbps compared to 96kHz Vorbis at 31kbps is about double the size and it sounds even worse (than Vorbis) (there is a lot of noise in the lower frequencies when a low

O'Connor, Kevin wrote: > Is this bleedover expected? I found an older copy of opus-tools (0.1.6, libopus 1.0.2) and it does not appear to > have the bleedover between channels, so I think this was introduced in 1.0.3 or 1.1. This bleed-over should not be possible with the CELT layer (_especially_ not during silence), but it is possible with the SILK layer. 1.1 added automatic speech

Hi! Excuse the delay, but I had to deal with a corrupted NTFS file system that ate many important files on an USB stick... The FLAC version of the original is almost 6MB and it can be downloaded slowly from this time-limited link: https://sbr5vjid0jgmce4q.myfritz.net:40262/nas/filelink.lua?id=0ba5a10529a6fe7b On the meaning of a logarithmic sweep: If you use foobar2000 and the

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

Occassionally on calls we get what sounds like low volume channel bleedover. Not clear enough to make out words, but not echo of either side of the main coversation. We're using a Digium card with 11 channels connected to PSTN lines. Any ideas on what the problem is or how to go about troubleshooting? Thanks! Paul

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. What sine sweep exactly? How did you obtain it, and how exactly did you encode and decode it? Jan > The strange > thing was that even though the output wave file was at 48 kHz, it

Hello Jean-Marc, yes, it works for both 32 and 48 kb/s. Also this option marginally increases the size of the file which is understandable. I tried a few other music files and found that this stereo narrowing happens not only in the case of track I submitted (although in smaller scale) so it would be great if the final libopus 1.2 had officialy a parameter to force music "mode" of

OK, so at the link : https://uloz.to/!yyVrCY2Y8sn1/devil-s-elbow-opus-7z (https://uloz.to/%21yyVrCY2Y8sn1/devil-s-elbow-opus-7z) (change the language to English by clicking at the flag at the right upper side of the web page or just simply click at "Stáhnout pomalu" - it may take some time as the file is 55 MB+ and the download service is free) there is 7zip archive with 5 music

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

On 05/11/2016 12:35 PM, Amit Ashara wrote: > I ran the opusenc.exe on a wave file and checked the OpusTag section. My > concern is on Total Segment Size being >> than the actual data being > put. Is this just an example of implementation or does a size of 764 > BYTES kept as a place holder for putting more data? Yes, opusenc does reserve some space in the header so that tags can

On 10/05/16 02:37 PM, Amit Ashara wrote: > Is there a format document on the OpusTag structure? Search always shows > up Vorbis but not Opus. The basic format is shared with Vorbis, but the 'magic signature' is different ('OpusTags' instead of '0x05vorbis') and vorbis puts a 0x01 value in an extra byte after the last tag. The OpusTag packet layout is described in

switch between speech and music encoding rather quickly. I wanted to try that for a CD track that has a vocal (talk) introduction = followed by music (just two instruments). Using foobar2000 with some custom setup for Opus, I tried VBR encoding = with a 256kbps setting (opus-tools-0.1.9-win32). My expectation was that = the bitrate would be significantly lower than 256kbps while speech is =

(Please wrap your lines.) On Oct 26 01:38:34, Ulrich.Windl at rz.uni-regensburg.de wrote: > Playing with Opus 1.3 I converted a tone sweep with a sample rate of 96kHz (just for fun). Before I had converted that from WAV to FLAC, and to Vorbis without problems. Can you please post the original wav? I am not sure what Audacity means by a logarithmisch sweep. Is that a fixed number of Hertz per

>>> Amit Ashara <ashara.amit at gmail.com> schrieb am 11.05.2016 um 19:32 in Nachricht <CAEyg9sjvTWMBMMCJ8HQcYmbv1BtNt54CgpqWaGNm02MWrKcxaQ at mail.gmail.com>: > Hello Jean-Marc, > > So for the moment we can assume that this method is also OK to use? > > On Embedded Systems, both SRAM and Flash can be a restricting factor > besides the compute time. To

Hello, I am trying to encode an Opus file with Ambisonics including Head-Locked (non-diegetic) Stereo sound for a Virtual Reality 360° video. YouTube describes the spatial audio requirements here: https://support.google.com/youtube/answer/6395969 It's the last list item 5. > 5. Supported First Order Ambisonics (FOA) with Head-Locked Stereo format: > W, Y, Z, X, L, R as a 6-channel

>>> Jan Stary <hans at stare.cz> schrieb am 05.11.2018 um 11:05 in Nachricht <20181105100534.GB44329 at www.stare.cz>: > (Are we off‑list now by intention?) No, just fooled by the list defaults (some need just reply, others need reply to all) > >> Did you also try to listen at the beginning, shortly before the real tone > appears in the audible spectrum?

Hi Amit, I'm not sure what you're trying to ask, but the reserved space in the header is an optional thing. Encoders are free to do it, but are not required to do so. Jean-Marc On 05/11/2016 01:32 PM, Amit Ashara wrote: > Hello Jean-Marc, > > So for the moment we can assume that this method is also OK to use? > > On Embedded Systems, both SRAM and Flash can be a

The overhead of Ogg (in file size) is pretty small and it's efficient enough for most applications (and uses far less CPU than the codec anyway). If anything, you might want to look at optimizing the existing Ogg implementation (e.g. like Tremor did in the context of Vorbis). Of course, you're always free to design a new container, but I doubt it's worth it and it's a lot of work

I have run a few more test at different bitrates and 1.1 is looking even worse in terms of speed compared to previous versions. I have shared a google sheet which has the raw data and charts for 6,16 and 32 kbps. Unfortunately you cannot show proper error bars on Google sheets but the standard deviation is in the data if you want to look. You can see that the profile for 1.1 is a lot different