similar to: Opus vs iSAC

The Opus RFC seems to recommend a frame size of 20ms for most applications. For wideband speech, the sweet spot range is recommended to be 16-20kbps. 20ms frames => 50 frames per second. For a VoIP application, the header overhead per frame (IP+UDP+RTP+SRTP) is 44bytes => 17.6kbps at 50 frames per second. So a 20ms frame size seems to cause a roughly 100% overhead of header data. Therefore,

Hi everyone, I'm happy to announce that Opus 1.3 was just released. This major release again brings quality improvements, new features, and bug fixes. You can read all the details in this release demo page: https://people.xiph.org/~jm/opus/opus-1.3/ Changes since 1.2.x include: - Improvements to the VAD and speech/music classification using an RNN - Support for ambisonics coding using

Hi everyone, I'm happy to announce that Opus 1.5 has just been released. Opus 1.5 is the first release to make extended use of ML in the encoder and decoder. You can read all the details in our demo page at: https://opus-codec.org/demo/opus-1.5/ Major changes since 1.4 include: - Significant improvement to packet loss robustness using Deep Redundancy (DRED) - Improved packet loss

And it turns out there was a missing meson file in the original release so here's 1.5.1 with the file added: https://downloads.xiph.org/releases/opus/opus-1.5.1.tar.gz Cheers, Jean-Marc On 2024-03-04 11:26, Jean-Marc Valin wrote: > Hi everyone, > > I'm happy to announce that Opus 1.5 has just been released. Opus 1.5 is > the first release to make extended use of ML in

Hi! So here is what I got with different encoder settings; still not sure what the "best" setting is. 6kbps seems to add distortions, so I tried 12kbps. MP3-Original: LAME 3.99r, 120kbps, 44100Hz, Stereo, VBR V5 (22:23, 19.8MB) Opus (--raw-rate 44100 --bitrate 56 --vbr --comp 5): (44:45?, 23.4MB): Broken Opus (--bitrate 56 --vbr --comp 5 --ignorelength - %d): (22:23, 12.1 MB, 74kbps)

FYI, the README file section on building for Windows says to see the README.md files in the cmake or meson subdirectories, but there are no README files in either of those subdirectories. -----Original Message----- From: opus <opus-bounces at xiph.org> On Behalf Of Jean-Marc Valin Sent: Monday, March 4, 2024 12:52 PM To: opus at xiph.org Subject: Re: [opus] Opus 1.5 is out -- now with ML

On 01/21/2015 07:51 PM, Daniel K wrote: > Hi guys, > > I'm using Opus for speech in wide-band mode (sampling rate 16000) and > 20ms frames with signal type set to SIGNAL_VOICE. > > I have a few questions here: > > 1. > When I choose VBR mode, the codec seems to choose the bitrate on its own. > However, that seems to be an issue on mobile devices. In some cases,

Hi guys, I'm using Opus for speech in wide-band mode (sampling rate 16000) and 20ms frames with signal type set to SIGNAL_VOICE. I have a few questions here: 1. When I choose VBR mode, the codec seems to choose the bitrate on its own. However, that seems to be an issue on mobile devices. In some cases, when I configure the bitrate to say 20kbps, I see that the outgoing codec bitrate at

On 7 Nov 2017 13:36, Lucas Clemente Vella <lvella at gmail.com> wrote: 2017-11-07 11:10 GMT-02:00 encrupted anonymous <sergeinakamoto at gmail.com<mailto:sergeinakamoto at gmail.com>>: did another test of many. NeroAAC q=1 @400kbps and Vorbis q=10 @412kbps shared 2nd place. OPUS @330 kbps - 3rd place. LAME MP3 q=0 @320 kbps - 1st place. ---JPEG file attached--- Please disable

On 2017-11-01, Jean-Marc Valin wrote: > I'm not sure, but my best guess would be "because MP3's window is very > leaky and MP3 has to waste a lot of bits in the LF because of that". > It could also be just the MP3 encoder being silly, or other things. Was the original poster speaking about the SILK or the CELT derived mode? Because at least wrt SILK (and the rest of

FYI, the Ogg Opus encapsulation is now RFC 7845: https://tools.ietf.org/html/rfc7845 -------- Forwarded Message -------- Subject: [codec] RFC 7845 on Ogg Encapsulation for the Opus Audio Codec Date: Fri, 29 Apr 2016 19:47:28 -0700 (PDT) From: rfc-editor at rfc-editor.org To: ietf-announce at ietf.org, rfc-dist at rfc-editor.org CC: drafts-update-ref at iana.org, codec at ietf.org, rfc-editor at

Hi Ulrich, Other than the bitrate, none of the discussed settings should have a significant impact on the file size as the encoder will always use the available bitrate to encode the input signal. The additional settings only configure where the encoder spends the bits and how much time it has to figure that out. >From my experience and as others noted, 15-32 kbps should yield very good

did another test of many. NeroAAC q=1 @400kbps and Vorbis q=10 @412kbps shared 2nd place. OPUS @330 kbps - 3rd place. LAME MP3 q=0 @320 kbps - 1st place. ---JPEG file attached--- Please disable speech synthezation in OPUS for 96 kbps and up. I don't want my music sound like from a phone speaker! Or what is the problem? Modern codec at high bitrates should produce nearly bit-exact sound, not

Hi, I just released Opus 1.3-beta. This is a beta release towards the upcoming Opus 1.3. Changes include: - The spec fixes in RFC 8251 are now enabled by default - Improvements to the VAD and speech/music classification using an RNN - Improvements to stereo speech coding at low bitrate - Added support for ambisonics projection using mapping 3 (disabled by default) - Fixes to the CELT PLC

Hi I was just looking at your Opus quality vs bit rate graph.? Latency looks good.? ? Two questions: 1/ Wideband: How did you address channel separation (i.e. What is the interaural time difference (ITD) granularity/resolution in microseconds? ?- assuming two channels. ) ? 2/ Narrowband: I see AMR-NB.? How does Opus compare with modern speach coding and band excitation models? I.e.

Hi, I have a question regarding the stereo capabilities of Opus. I would like to establish a connection between two ends via Wi-Fi and the signals that are to be transmitted are of speech kind. It mean on both ends speech is both recorded and played back as stereo. Now would the delay and loss characteristics of the speech transmission at a certain bitrate be the same as Mono voice transmission?

Hi folks sincere apologies if this is not meant to be posted here. I am the CEO at http://directi.com. One of our products http://talk.to is in the messaging space and we are planning to add voice support to our application and are currently investigating codecs (Opus, ilbc, isac, G.729, AMR etc). There are several variables and obviously Opus would be our preferred choice given its quality, VBR,

Hi folks sincere apologies if this is not meant to be posted here. I am the CEO at http://directi.com. One of our products http://talk.to is in the messaging space and we are planning to add voice support to our application and are currently investigating codecs (Opus, ilbc, isac, G.729, AMR etc). There are several variables and obviously Opus would be our preferred choice given its quality, VBR,

Hi I am Rupesh from India. I have a huge directory of size 93.5 gb with 8500 mp3 files and 2000 sub directories. All these mp3s are speeches recorded by someone at 64 kbps. I want to compress these files recursively to opus using lame or another tool with 16 kbps bit rate and 11050 sample rate. I have compressed the above huge directory with above options using ffmpeg and the resulted

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