similar to: Opus 1.2 released!

Hi everyone, I've just released Opus 1.2-alpha. It comes with many improvements, including: - Speech quality improvements especially in the 12-20 kbit/s range - Improved VBR encoding for hybrid mode - More aggressive use of wider speech bandwidth, including fullband speech starting at 14 kbit/s - Music quality improvements in the 32-48 kb/s range - Generic and SSE CELT optimizations - Support

Hi everyone, Just to let you know that Opus 1.2 is finally here. This major release brings many quality improvements, new features, and bug fixes. You can read all the details (and hear audio samples) in this release demo page page: https://people.xiph.org/~jm/opus/opus-1.2/ Changes since 1.1.x include: - Speech quality improvements especially in the 12-20 kbit/s range - Improved VBR encoding

I know the focus for Opus is low delay, but I've been watching its development with interest because of the potential for audiobook/podcast use, where latency is practically irrelevant. I hear the upcoming USAC codec will give good results for this niche (though listening test results don't seem to be available to the public yet), but I also hear it'll be extremely patent

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi to everybody. First of all I hope this is the right place to discuss such an (nitpicky) issue. I've just been testing the current Opus release and for mere curiosity compared its performance to WMAPro with CD quality music at low bitrates (48 kbit/s). While Opus generally does a very good job, I found one particular example (a high pitched

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

Thank you Mark. I agree and have now updated the pull request with a new commit, addressing your comments. Please take a look. /Gustaf On Fri, 5 Apr 2019 at 11:41, Mark Harris <mark.hsj at gmail.com> wrote: > On 2019-04-01 3:37, Gustaf Ullberg wrote: > > Hi everyone, > > > > Some time ago, I sent a pull request > > <https://github.com/xiph/opus/pull/107>

Thanks Jean-Marc, I was able to get both encode and decode working the CC3220 device! But for bi-directional communication, I need decode and encode to occur in less time than the frame size I’m sending (20 ms). Currently decode takes 16~22 ms and encode is ~13 ms. What is the best way to try to reduce this time? Also, unsure why encode is taking less time than decode... I've also

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

Hello everyone, I have been having trouble utilizing the PLC function (I think). In older versions (0.7.1) a pointer to NULL did the trick, but with code from the tip of the GIT repository, this method causes the codec to crash on the C55x fixed point platform. I have not determined where the crash occurs. I have also attempted to pass in an array of zero's and this creates an echoey, reverby

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

Good morning. I've ran a test against MP3 format. Code: (first convert tested audio file to 16 bit 48khz with sox.exe if needed) lame.exe -b 320 48khzfilein.wav -o fileout.mp3 lame --decode fileout.mp3 -o fileout.mp3.wav opusenc.exe --bitrate 320 48khzfilein.wav fileout.opus opusdec.exe fileout.opus fileout.opus.wav wavdiff.exe 48khzfilein.wav fileout.mp3.wav -diff fileout.mp3.delta.wav

Using -i should prevent failing if the files don't exist. --- appveyor.yml | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/appveyor.yml b/appveyor.yml index c85b0b1..ad9c6c0 100644 --- a/appveyor.yml +++ b/appveyor.yml @@ -17,7 +17,7 @@ build: verbosity: minimal after_build: -- cmd: 7z a opus.zip win32\VS2015\%PLATFORM%\%CONFIGURATION%\opus.lib include\*.h +- cmd: 7z

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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

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 all, after FOMS I decided to take a look at the opus library and I found that I could improve a bit the build system and cleanup the code a little bit. Most of the changes to the code has been suggested by my two tools cowstats and missingstatic (part of the ruby-elf gem if you care). HTH, Diego

Jean-Mark sarkasm. Jean-Markasm. (Bonus points for providing an actual noisy WAV! ^_^) On 30/10/2017 20:28, Jean-Marc Valin wrote: Hi, Before I comment on the graphics you posted to visualize the difference between two audio signals, I'd like to ask for your help in evaluating my JPEG encoder. I've encoded an image with JPEG and then computed the difference with the original. I then

Hi guys, as MP3 and Opus have very similar objectives, I think the original poster's question was a valid one: Why does Opus have more artefacts in the lower frequency ranges than MP3 has? The spontaneous suspect that lower frequency artefacts may be more noticeably than higher frequency artefacts seems plausible, also. Is it a matter of energy (which is higher for higher frequencies)? When

Hello Thomas and Amit, Thanks for your notice and the detailed decode performance report. I describe the details of my encode/decode test on STM32F407ZG. A. opus version: latest 1.2.1 (TI: opus 1.1.2) B. KEIL 5.23 (TI: ARM compiler tool chain 5.2.7) C. setup the encoder as the below (fs is the sampling frequency) enc = opus_encoder_create(fs, chans, OPUS_APPLICATION_AUDIO, &opus_err);

I?m trying to build Opus (git master) for iOS, and it doesn?t build unless I disable the ARM assembly. It looks like the problem is that Apple?s assembler doesn?t support all the assembler directives that the GNU assembler does. I suspect this is a combination of the fact that Apple platforms are Mach-O rather than Elf, and just the fact that Apple?s assembler is extremely divergent from the