similar to: Opus 1.2.1 released

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

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, 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 converted the difference to sound. You can listen to the image difference on this clip: https://jmvalin.ca/misc_stuff/diff.wav Can you

Just to be clear, my goal here wasn't to make fun of anyone, but to drive the point that spectrograms should *never* be used to demonstrate quality. The only case where they can sometimes be useful is for diagnostic purposes. If you hear something and you're not sure what you're hearing exactly, then sometimes a spectrogram can help you figure out what it is. That's pretty much it.

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

> 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

Hello, I understand it would be better to post several messages with separate topics but I hope I don't cause too much mess if I put it all in a single message this time. To be clear, recently I've been testing Opus Tools under Windows and these are my questions/observations. ????#1. To test encoding at low bitrates, I encoded a sine sweep at 12 kbps with Opusenc and then decoded

On Aug 07 22:04:21, petrparizek2000 at yahoo.com wrote: > > The encoded opus file is 48kHz, > > so how would the output wav be resampled from 16kHz? To be clear: did you mean the opus output of opusenc or the wav output of opusdec? > > What are those "clear signs" exactly? > > The things that I can hear while listening at 1/2 or even 1/4 of the > original

> Why are you using a stereo file > containing the same sweep in both channels > and then downmixing to mono? When I first tried encoding at a higher bitrate, I needed to test the different behavior of the "mid" (l+r) and "side" (l-r) channels. That's why I made the first sweep identical on both the left and the right channel (i.e. "side" is silent)

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

Hi Julien, Quote from : http://dspguru.com/dsp/faqs/multirate/resampling "The problem is that for resampling factors close to 1.0, the interpolation factor can be quite large. For example, in the case described above of changing from the sampling rate from 48 kHz to 44.1 kHz, the ratio is only 0.91875, yet the interpolation factor is 147!" My guess is that Opus would perform similar to

> To be clear: did you mean the opus output of opusenc > or the wav output of opusdec? I meant during the decoding. There's one significant difference between how Opusdec deals with resampling and how, let's say, MP3 decoders usually deal with resampling. If I make an MP3 at a very low bitrate and if the encoder decides (because it's too low) to internally resample my audio

Any luck reproducing the problem with opus_demo or opus-tools? Jean-Marc On 02/22/2018 10:14 PM, Stepan Salenikovich wrote: > > > On Thu, Feb 22, 2018 at 9:53 PM, Jean-Marc Valin <jmvalin at jmvalin.ca > <mailto:jmvalin at jmvalin.ca>> wrote: > > On 02/22/2018 09:34 PM, Stepan Salenikovich wrote: > > Its unexpected because the decoder continues to

Hi, another question on the same topic Speex resampler at 44.1kHz seems to be very CPU intensive on Android (even more than the Opus encoder) While Speex at 48kHz is just fine. I wonder any alternate solutions or ideas ? Improve it, look for alternate solution ... I am guessing the NEON optimization are still used for both, etc. On Thu, Apr 2, 2015 at 4:46 PM, Jean-Marc Valin <jmvalin at

On Thu, Feb 22, 2018 at 9:53 PM, Jean-Marc Valin <jmvalin at jmvalin.ca> wrote: > On 02/22/2018 09:34 PM, Stepan Salenikovich wrote: > > Its unexpected because the decoder continues to output all samples > > of -32768 even when the microphone input is silence or near silence, so > > I would expect the decoded values to be at or near 0. > > Oh, if the output is

What is the possibility to use the Cortex-M4 DSP instructions to fully optimize the OPUS code? Could we use the ARM CMSIS DSP library for this optimization? Thanks, Heng -----Original Message----- From: opus-bounces at xiph.org [mailto:opus-bounces at xiph.org] On Behalf Of opus-request at xiph.org Sent: Tuesday, November 04, 2014 2:00 PM To: opus at xiph.orgis Subject: opus Digest, Vol 70,

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

On Thu, Feb 22, 2018 at 8:34 PM, Jean-Marc Valin <jmvalin at jmvalin.ca> wrote: > Hi Stepan, > > I would need more information to be able to investigate further. It's > legal for the decoder to output -32768, so it would be good if you could > explain how this is unexpected. Its unexpected because the decoder continues to output all samples of -32768 even when the

Hi, I work on a webRTC application and recently tried updating from opus 1.1.5 to 1.2.1 Afterwards I noticed occasionally weird audio glitches. I finally tracked down the issue to the opus decoder in my application outputting samples with the value of -32768. This behaviour stopped when reverting to opus 1.1.5 or compiling opus 1.2.1 without configuring --enable-float-aprox and --0fast. The

Just to make sure, what's the goal here? Is the goal 1) to have a fast Opus implementation or are you 2) looking for an interesting FPGA implementation project? If 1), then an FPGA is most likely not necessary since Opus is not computationally expensive. If 2), then it depends on the desired size of the project and the desired quality. The simplest encoder possible is indeed simpler than the