similar to: Understanding libopus parameters

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

Hi Jean-Marc, A simpler question. How does opus_encode() generate packets of 20ms (SILK-only or Hybrid)? Concatenating two 10ms frames or doing it straight with just one 20ms frame?

>On 11/20/2013 03:37 PM, O'Connor, Kevin wrote: >> I have an application I intend to use Opus for that involves creating >> recordings of two-party conversations where each party is saved as a >> separate channel. Audio may be later processed or analyzed on a >> per-party basis so if audio in one channel affects the other channel, it >> could create problems.

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

The libopus encoder's opus_encode() method is documented as "Encodes an Opus frame". Does that mean that it always produces a single Opus frame (i.e. the number of frames in the TOC byte will always be 0)? It's not clear from the documentation, but the fact that it cannot produce a 120ms Opus packet makes me wonder if that was the intention and any multi-frame Opus packets must

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 10/26/2013 01:11 PM, Wang, Chris wrote: > A simpler question. How does opus_encode() generate packets of 20ms > (SILK-only or Hybrid)? Concatenating two 10ms frames or doing it > straight with just one 20ms frame? Just one 20 ms frame. It always returns a single frame except when it just can't (e.g. 60 ms CELT). > From your explanations below, opus_encode() will concatenate

Thanks Jean-Marc and Benjamin for the answers. One follow-up question. If I use a repacketizer as Jean-Marc suggested by combining two 60ms frames to form a 120ms frame, without extracting individual frames and using a new TOC, I would need to have a "de-packetizer" that does the exact opposite of repacketizer. De-packetizer would need to separate this 120ms frame into two 60ms frames

Hi, I'd like to know the reasons why CELT supports only signals with sampling frequency in the range of 32-96 kHz. In effect, it can clearly outperform speex at high bitrates, and has potential to be used in high quality voice communications even for 11, 16 and 22 kHz speech signals. It could also compete with SILK codec (to be soon released by Skype). See this page for more specifications

Hi Peter, There's two main issues with a patch like the one you're proposing. First, the data is only valid when SILK is being used and is essentially undefined in CELT mode. The second issue is that by exposing internals, it makes it impossible to improve these algorithms since it would break API compatibility. I'm not fundamentally against trying to expose some information, but

> As the thing is to encode for human ears (AFAIK), I'd say that 4kHz is already "quite high", > and I wonder who can actually hear pure 20kHz sine. If you read the beginning of RFC 6716, you learn that Opus never encodes any frequencies that are higher than 20 kHz. So at some medium or high bitrates, anything above 20 kHz is filtered out, not because of the bitrate but

Hi all! What is the right way to report bugs in libopus? I couldn't find any publicly available bugtracker. Anyway, the problem I need to report is that some source files in libopus are missing copyright headers. This breaks license check script that we use in chromium to validate licenses for third-party dependencies. Specifically the following files don't have copyright header:

I have just started trying Opus with a view to using it in a project. I am interested in embedded hardware and tried it on the Raspberry Pi using the raspbian distro. The version of libopus in the repos is 0.9.14. I installed this and tried encoding 2 minutes of speech from a librevox recording. It managed this at a respectable pace for complexity 10: Skipping chunk of type "LIST",

Hi Peter, Can you say a little bit more about what you're doing exactly with the information you're exposing and how? unfortunately, I don't have a concrete proposal in mind right now. That's in part because I don't quite understand the use case, but also because it's really hard to expose this kind of information in a way that both avoids breaking application with new

Hello, I'm not a programmer or Opus developer but I tried to test the sound quality of a music file (Nick Warren - Devil's Elbow, freely available from the author's Soundcloud account) encoded with libopus 1.2 RC1. I used Windows binaries from free-codecs.com. I noticed that in the case of my selected music file (which is generally harsh on lossy codecs as it's necessary to

hi, I am porting opus encoder to a low end device with 32K ram, 256K flash and 32MHz arm M3 mcu. But opus seems consume too much. To make it work , what I can think of 1, Only fixed point supported 2, Only mono voice application supported 3, Set complexity to zero 4, Support only one sample rate, like 16KHz 5, Silk mode only or Celt mode only My question is , before

Hello, I received feedback from engineers working on NE10 [1] that it would be better to use NE10 [1] for FFT optimizations for opus use cases. However, these FFT patches are currently in review and haven't been integrated into NE10 yet. While the FFT functions in NE10 are getting baked, I wanted to optimize the celt_pitch_xcorr (floating point only) and use it to introduce ARM NEON

I was digging around and saw a bunch of different people have created ways of building libopus for android (even I did one back in the CELT days). Is there an official one somewhere that I missed? I even notice this is the Google sources, but it seems to be a touch long in tooth at this point: https://android.googlesource.com/platform/external/libopus/+/android-wear-5.1.1_r1 I am happy to build

Hi everyone, I've made several posts recently about CELT being "replaced" by the Opus codec ( http://opus-codec.org/ ) and I thought it was time to give an update on what's going on. As many of you know, I've been involved at the IETF on this new Opus codec, which essentially merge (a modified version of) Skype's SILK codec with CELT. This is more than just two codecs

> > I am talking about the original sweep. > > The original sweep stops pretty close to 24 kHz. I mean the original sweep _as_encoded_, sorry.