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Old 18th July 2013, 18:12   #1  |  Link
IgorC
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Opus 1.2.1/1.3RC2. A new WebM audio codec (Opus+VP9/AV1)

Hello,

It's a release of Opus 1.1 (UPDATED: and new 1.2.1)




Demos
(version 1.1)http://people.xiph.org/~xiphmont/demo/opus/demo3.shtml
(version 1.2) https://people.xiph.org/~jm/opus/opus-1.2/

Previously Opus has shown excellent results beating LC-AAC, HE-AAC and Vorbis.

Public Multiformat Listening Test @ ~64 kbps [March 2011]
Public Multiformat Listening Test @ ~96 kbps [2014]

Official builds of Opus 1.2.1

Win64 https://ci.appveyor.com/api/buildjob...opus-tools.zip

Win32 https://ci.appveyor.com/api/buildjob...opus-tools.zip

You can also find some custom builds here
https://hydrogenaud.io/index.php/top...html#msg941283

Playback

You can play Opus files on your Android/Apple iOS/Windows Phone smartphones/tablets via foobar2000 http://mobile.foobar2000.com/

Many players as MPC, VLC, foobar2000 (Windows, Android, iOS, Windows Phone) support Opus format as well as Rockbox (a firmware for portable players). https://www.rockbox.org/

For more information
http://wiki.hydrogenaud.io/index.php?title=Opus

Last edited by IgorC; 20th September 2018 at 18:08.
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Old 6th December 2013, 16:36   #2  |  Link
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Opus v1.1 has finally been released

For info see here:
http://xiph.org/~xiphmont/demo/opus/demo3.shtml

Fresh builds can be found here:
http://sourceforge.net/projects/muld.../Opus%20Tools/
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Old 7th December 2013, 04:23   #3  |  Link
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Have updated the OP and links to official binaries.
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Old 7th December 2013, 14:51   #4  |  Link
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What do people think about Opus's surround sound encoding capabilities?
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Old 9th December 2013, 10:15   #5  |  Link
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Quote:
Originally Posted by LoRd_MuldeR View Post
I've tried your SSE2 x86 build and compared it with the official build from Opus-tools v0.1.8.
As expected, your build is a bit faster than the Xiph one, however it writes 60 bytes less when encoding the same source file on the same machine:

Xiph:
Code:
Encoding using libopus 1.1 (audio)
-----------------------------------------------------
       Encoded: 44 minutes and 53.6 seconds
       Runtime: 1 minute and 24 seconds
                (32.07x realtime)
         Wrote: 43510600 bytes, 134680 packets, 2696 pages

Mulder:
Code:
Encoding using libopus 1.1-SSE2 [2013-12-06] (audio)
-----------------------------------------------------
       Encoded: 44 minutes and 53.6 seconds
       Runtime: 1 minute and 21 seconds
                (33.25x realtime)
         Wrote: 43510540 bytes, 134680 packets, 2696 pages
Perhaps you can find some usefull info about that here:
http://www.hydrogenaudio.org/forums/...dpost&p=851264
http://www.hydrogenaudio.org/forums/...dpost&p=851337
http://www.hydrogenaudio.org/forums/...dpost&p=851357
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Old 9th December 2013, 11:07   #6  |  Link
the_weirdo
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@Brazil2

According to Jean-Marc Valin, that may be due to rounding behaviour (with different optimization levels).
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Old 10th December 2013, 19:57   #7  |  Link
Anakunda
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I would be interested how low can man go to keep transparent sound. Seems that effective ~210k still give superior quality on 6 channel audio.

Quote:
surround with good quality at 128 kbps for 5.1 and usable down to 48 kbps
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Old 10th December 2013, 21:53   #8  |  Link
LoRd_MuldeR
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Quote:
Originally Posted by the_weirdo View Post
According to Jean-Marc Valin, that may be due to rounding behaviour (with different optimization levels).
Either that, or the phase of the moon. 60 bytes more/less on a 43510600 bytes file is a difference of 0.00014 percent - which is pretty much nothing

Not that a difference in file size would mean anything. In theory, the bits could be distributed completely different and still the exactly same file size could be hit!

Instead, you should decompress both files to uncompressed Wave/PCM and then add the inverse of the one file to the other file (e.g. use "mix paste" in Audition).

As long as the result is silent - or at least very close to silent - there is no reason to be concerned. Furthermore, you could also try to ABX the files...

[EDIT]

I just converted a complete album (Chinese Democracy) to Opus format, once using the "i386" build and once using the "sse2" build. Both files came out at identical size.

Files were not identical by content. Of course, not. The encoder signature string already differs.

Next I decoded the "i386"-encoded file with the "sse2" decoder and the "sse2"-encoded file with the "i386" decoder, which should be the most extreme test case, I think.

The difference between the two decoded files is silent. There are a few minor peaks in a few places, but noting critical. Average RMS of the diff is -85.79 dB.
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Last edited by LoRd_MuldeR; 11th December 2013 at 00:33.
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Old 11th December 2013, 08:08   #9  |  Link
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Actually, I've discovered that if you compile Opus and Opus-tools with GCC, you can add "-fno-strict-aliasing" to CFLAGS to get the same (bit identical) results with -O level from 0 to 3. (Of course, it only applies when you're comparing the results encoded by the same machine.)
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Old 13th April 2014, 14:22   #10  |  Link
LoRd_MuldeR
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FWIW, here's a set of fresh Opus binaries made from current Git Master:
http://sourceforge.net/projects/muld...3.zip/download
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Old 14th January 2015, 15:38   #11  |  Link
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fyi, Opus Audio format has been validated by mp4ra.org...and a draft is in progress to encapsulate Opus files into MP4.
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Old 14th January 2015, 16:17   #12  |  Link
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Quote:
Originally Posted by Kurtnoise View Post
fyi, Opus Audio format has been validated by mp4ra.org...and a draft is in progress to encapsulate Opus files into MP4.
I've must admit I was not expecting this move. But I'll take it
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Old 14th January 2015, 17:13   #13  |  Link
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Let's see how much time the GPAC team will take to implement the idea...

since they don't care at all about VC-1 and DTS in MP4
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Old 14th January 2015, 17:48   #14  |  Link
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Let's see how much time the GPAC team will take to implement the idea...
http://sourceforge.net/p/gpac/feature-requests/59/
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Old 15th January 2015, 08:38   #15  |  Link
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Quote:
Originally Posted by Kurtnoise View Post
fyi, Opus Audio format has been validated by mp4ra.org...and a draft is in progress to encapsulate Opus files into MP4.
Also, this has been added in vlc few hours ago...
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Old 15th January 2015, 10:24   #16  |  Link
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Opus in MP4 is still a draft, and VLC adding a potentially unfinished draft just seems like a not-well-thoughtout move. At least they only demux and don't mux files which may be wrong eventually...
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Old 27th January 2015, 15:39   #17  |  Link
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Not sure if this a a playback or a 'I'm just stupid' king of bug.
I'm using:
Code:
ffmpeg -y -threads 8 -loglevel fatal -i "H:\Output\test.ac3" -ac 6 -ar 48000 -f sox - | sox --multi-threaded --ignore-length --temp "H:\Temp" --buffer 524288 -S -t sox - -b 16 -t wav - | opusenc --bitrate 192 --comp 10 --framesize 20 --expect-loss 0 --max-delay 1000 --ignorelength --raw-bits 16 --raw-rate 48000 --raw-chan 6 - "H:\Output\test.opus"
to convert an .ac3 into an .opus file. (uploaded both to: https://drive.google.com/folderview?...ms&usp=sharing)
Reencoding seems to work fine, but upon playback the channel order is wrong lfe comes through the front right box.
-> My guess is that I either need to rearrange the channels before encoding or there is a problem with the decoding through MPC-HC.
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Old 27th January 2015, 15:51   #18  |  Link
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Quote:
Originally Posted by Selur View Post
Reencoding seems to work fine, but upon playback the channel order is wrong lfe comes through the front right box.
-> My guess is that I either need to rearrange the channels before encoding or there is a problem with the decoding through MPC-HC.
That's right the order in Ur conversion is messed.
Try play this: http://www6.zippyshare.com/v/50RpBiHm/file.html

eac3to test.ac3 stdout.wav -normalize | opusenc --ignorelength --vbr --comp 10 --bitrate 192 - test.opus

Last edited by Anakunda; 27th January 2015 at 15:55.
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Old 27th January 2015, 18:30   #19  |  Link
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Quote:
Originally Posted by Anakunda View Post
That's right the order in Ur conversion is messed.
Try play this: http://www6.zippyshare.com/v/50RpBiHm/file.html

eac3to test.ac3 stdout.wav -normalize | opusenc --ignorelength --vbr --comp 10 --bitrate 192 - test.opus
I can confirm that the 'LAV Audio Decoder' (v0.63.0) plays/presents the channels in the correct order

EDIT: The last time I checked, LameXP was able to correctly assign the channel order...
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Last edited by SeeMoreDigital; 27th January 2015 at 18:34.
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Old 27th January 2015, 19:10   #20  |  Link
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Okay, then ffmpeg probably doesn't adjust the channel order is '-f sox' is used.
Thanks for the feedback.
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