Vorbis Opus Mp3 Comparison
What's with the FUD, dude? Pre-release Opus was included in the hydrogenaudio listening tests months ago alongside all the relevant (not shit) implementations of HE-AAC. Just because it doesn't say 'plus' doesn't mean it doesn't include SBR (or PS or any of the other bells and whistles).SBR is good for perceptual 'niceness', but as far as sounding like the original goes, it's often quite harmful. All these things are accounted for with the hydrogenaudio listening tests - they're an extremely anal community and wouldn't dare let prejudice through without a long and protracted flamewar. They're actually very trustworthy.
Vorbis, the audio format, can be used with other container formats (Matroska) and video, and still retains all advantages to MP3. The encoding itself in Vorbis is more efficient than MP3, due to it being 15 years younger. A 128k constant rate Vorbis file will always beat a 128k constant rate MP3 file. Actually, ogg is a magic format, I always ABX ogg@192vbr(.6) vs flac and the ogg vorbis file has a very gentle rolloff that makes it sound slightly less harsh than the original file MP3 at V0 is not as good IMO. Mp3 compresses the waveform a little compared to ogg. You can see it by looking at the waveform on audition.
Vorbis Vs Aac
Aacplus was just the early CT-proprietary version of HE-AAC. They did test against the two best publicly available HE-AAC encoders, which have improved quite a bit since the aacplus days. xiph.orgOpus has band folding, which is in some ways similar to SBR but considerably superior.
Halfway down Monty's two-year-old xiph.org there's some explanation and a visual of what this looks like on a spectrogram in low-bitrate situations. (Opus used technology from CELT but is considerably improved.)If you really think HE-AAC type codecs sound like CD at 32kbps and so forth you are extremely insensitive to coding artifacts. Unless you meant mono for all of those. The point is to be able to use lower bitrates and get the same quality. This is especially useful for things like audio streaming over the internet, where less bandwidth used equals more space for listeners.For audio streaming over the Internet, it's even more important to gracefully deal with packet loss and packets arriving out of order.
The point is what it's always been - streaming audio over the internet. Sure, FLAC is fine for LAN use.The good: this does (in theory) AAC quality compress, but with much lower latency.
A use case: playing music live over the internet. You need high quality, but if the latency is over 1ms, every musician is compensating for the delay already, and if it's over about 10ms, you just can't play together. And since it's music, the quality does matter.Another good thing, the XKCD cartoon not withstanding, is tha. What would make an audio codec something worth using that would make you switch?I would assume that widespread support among major applications would be an issue. You could also throw in the ability to compact an audio stream better than alternatives might be useful in some applications. Simply having content in that codec would be very useful as well.I would say being patent and license free (aka it can be incorporated into a GPL'd application) would be pretty far down the list, but not needing to pay a licensing fee might make the difference for some marginal applications or for start up groups needing some sort of audio playback where even a few extra dollars in royalties can end up costing more than it is worth (such as is the case for the current MP3 format).Then again that is sort of what pushed the VHS format over Betamax in the video tape format wars.
Small independent producers could mass produce VHS tapes cheaper than the Betamax tapes, and for marginal videos (.cough. porn movies.cough.) that made all of the difference.The problem here is that audio codecs are pretty entrenched and as you've suggested that even free alternatives are available. Unless there is something substantially different being done by this codec that even a non-techie can notice and suggest that this new algorithm is substantially better, I really have a hard time seeing this being adopted widely. There might be some niche applications if the compression algorithm is even a few percentage points better, such as perhaps a transmission protocol for audio on the Iridium satellites. Something like that may even be useful to have an on the fly codec converter depending on how it is used. My 5g ipod with rockbox did between 10 and 11 hours playing -q4 (128kbps) vorbis last time I tested it, which was really at least 3 years ago, probably better now.
I know they've done improvements to the codec implimentation since then.the same 5g ipod playing similar bitrate MP3, tested around the same time, was better by about 2 hours.I'm not sure what qualifies as 'guzzling', but I doubt I'm ever going to listen to my ipod for more than 11 hours without recharging it, especially considering I keep it pl. There is no dominant format at the moment. Music is ogg, mp3, flac and probably a few others. Flac is loosless, so it won't dissapear, but the other two gradualy will.The html5 tag hasn't been used much yet, and I'm betting +Opus will be the one to domainte over current flash-only players (since it seems it'll be the best supported format).Movies in MKV files are actually container with video streams and audio streams. There's also a small variety of formats used for those audio streams, and maybe Opus catches on. I certainly hope it does.But the market is fragmented, there's lots of different format being used in different areas.
Opus has a lot of giants behind it, if they do their part, Opus support will be better than that of many other formats in the long run, hence users will tend to adopt it, in time. In Firefox 15 (the current version) already added support for Opus.Opus is one of the 2 audio codecs which are mandatory-to-implement if a browser wants to support WebRTC (real time communication: video chat, voip from the browser and all that jazz).Telco's are following at WebRTC really closely, some see it as an oppertunity. Other probably not.So your smartphone might be getting support for it soon.So will your fancy TV in the near future include a browser? And a webcam (some already do)? And because of. Using Vorbis as an example, it's actually commonly used in a number of applications (like video games) where they don't want to pay licensing fees for every copy sold.
Unfortunately, this doesn't translate very well to consumer usage. People paying for music are getting it in AAC, and people downloading it are getting it in MP3. Transcoding the audio is essentially a loss no matter what.If Windows, Mac, and Android all began including the codec automatically then you could potentially see quick uptake.
Then again that is sort of what pushed the VHS format over Betamax in the video tape format wars. Small independent producers could mass produce VHS tapes cheaper than the Betamax tapes, and for marginal videos (.cough. porn movies.cough.) that made all of the difference.Sony actually refused to grant porn producers a licence to sell Betamax tapes, which is why the porn producers used VHS. So it wasn't about VHS being cheaper, it came down to the fact that they were simply not allowed to use Betamax.
Interestingly, when BluRay first appeared (also a Sony format), Sony again refused to licence it to. Actually, I belive this one might be the exception.
So many mayor players major playes have participated and are standing behing Opus, I can easily see this becoming the dominant codec for loosy audio. It won't displace flac, as flac is looseless, but it will displace oga, mp3, and other major players given time.I'm pretty sure it'll become the de facto standard in web as well, given the browser support, and HTML5's new tag.(I know that XKCD comic is meant to be a joke, but it does actually prefectly reflect what happens with almost every new standard these days). It can support up to 255 channels. The two-channel maximum is per stream. Multiple streams can be packed into single frames, but for 2 channels the mapping and coupling has to be signaled at the container level.The standard tools available at opus-codec.org use Ogg as a container for 'at-rest files,' and Firefox, foobar2000, and gstreamer-supporting apps (like Opera on Linux) all play Opus-in-Ogg files. VLC and Rockbox will soon release versions with playback support for these too. Though RTP etc is a primary focus, the 'at-rest file' support is ahead of the interactive support at this stage of the game.A Matroska mapping is still in progress.
Most likely, for the time being, Opus files will be predominantly Ogg, while the Matroska mapping will be more important for using Opus with video streams (esp. Vp8, improving on the webm vp8+vorbis+matroska combination). Seems to cover a wide range of range applications. I wonder why they left out loseless encoding. That would have made it the one true codec for everything.A quick look at the graph shows that they stop at 128kbps, which would mean it's a great codec for high-quality real-time audio telephony rather than as a codec to span the spectrum of low end real time to lossless audio.At least looking at the page - the summary mentions it's the 'one codec to rule them all', but the page leads me to believe it's somethi. Testing that things work has been done for all kinds of bitrates (512kbps per stream.
multiple streams in a surround encoding). It's just that Opus is transparent to most listeners on most samples before you hit 128kbps for stereo.
It's extremely hard to do a worthwhile listening test when only a few listeners can tell even a few of the samples from the original.Some people at hydrogenaudio.org have reported problem samples which they were able to ABX from the original at up to 160kbps. I haven't personally found any stereo samples I can reliably ABX from the original at above 80kbps.Of course lossless has its place. You don't want to be doing a lot of decoding lossy files, editing them, and then re-encoding, since you'll get wikipedia.org. A quick look at the graph shows that they stop at 128kbps, which would mean it's a great codec for high-quality real-time audio telephony rather than as a codec to span the spectrum of low end real time to lossless audio.The reason the graph stops at 128 kb/s is that things become uninteresting at that point - because nobody's able to actually tell the difference. With VBR, we've never had anyone report audio not being transparent above 200 kb/s.
There's a reason people don't want to organize listening tests at 128 kb/s and (especially) above. It's indeed the case that we don't support lossless. That one is already covered very well by FLAC and there was no point adding completely different algorithms to handle that.
Otherwise, Opus can replace MP3/AAC/Vorbis at rates above 128 kb/s too. 'Shine' is a really funny word for what HE-AAC sounds like at 16-24kbps. You can't polish a turd.As far as AMR-WB/NB, you have to get down to 8kbps before AMR-WB sounds measurably better, and you have to get down to 6kbps before AMR-NB sounds better. Opus is tied with AMR-WB at 12kbps and better at 16kbps, and it's tied with AMR-NB at 8kbps and significantly better at 12 or above. Look at the studies linked from the opus-codec.org if you want more details, keeping in mind that the Opus encoder has continued to improve in the year since those studies were done.
It's true that Opus does better than AAC and Vorbis at CD-quality bitrates and thus would be an improvement for music players etc.But the improvements there are fairly small- in fact, Opus wasn't originally targeted at that kind of use at all, and the authors were quite surprised that it outdid those kinds of high-latency codecs. Opus is a very low-latency codec, and it combines Skype's speech compression technology and more music-oriented technologies (those introduced in CELT) to allow interactive speech and music over the Web.Gaining marketshare in the high-bitrate stored music market against dominant formats like MP3 and AAC is hard, even when you outperform them substantially. But there's not really any established players in low-latency Internet audio. Opus blows all the other low-latency and/or low-bitrate codecs out of the water when competing in those other codecs' bitrate-latency 'sweet spots', is the only codec which can compete across that kind of a range, is now a standard, is royalty-free, and is already implemented in Firefox.Those who are saying 'meh, only audiophiles will care' or 'this won't get marketshare against AAC' are missing the point.
This codec will change the face of the Web. You're confusing transmission latency with algorithmic latency. If you're encoding music to store on an mp3 player, the format can use larger transform windows (usually MDCT) and other methods which mean the encoder looks at a larger number of samples before sending any output and the decoder has to read a larger amount of data before outputting any audio.For codecs like mp3, AAC, etc, even if you had an infinitely fast computer and infinitely fast transmission, adding an encoder and decoder between the recording and the playback adds 200ms of latency. That's fine for storing files on an mp3 player but totally unacceptable for real-time communication.
Opus, by default, has 20ms of algorithmic latency, and it can be configured to go as low as 2.5ms. No, he is using them correctly, referring to the application, not the medium.Music playback doesn't require low latency; it doesn't matter if there is a 500ms delay between when you press the play button and you hear the music. Because of this data is encoded in (relatively) large blocks to allow for as much compression as possible.VoIP on the otherhand does require low latency (100ms max). Otherwise it is very difficult to carry on a conversation because otherwise if you were to speak during a silence, it may not still be silent when the signal gets to when the other person, so you constantly talk over one another.
The other potential application, live interactive music, requires even lower latency for musicians to keep in sync with each other. For this reason Opus is designed to encode in small blocks of data to obtain better latency. When applied to a codec, 'latency' (obviously) refers to stream latency, not network latency (the latter has nothing to do with a codec, obviously).
The problem with codecs like MP3 for streaming purposes is that they encode fairly large 'frames' of audio, and these frames must be recorded before they can be encoded, encoded before they can be transmitted, received before they can be decoded, and quite possibly also decoded (fully) before they can be played. It may be possible to begin playing before the decoding is complete, which would help a lot, but it also might not - it depends on the codec.Suppose you've got a 'high latency' codec (such as MP3) that uses a 250ms frame and requires full decoding (this is an example; I don't know the actual numbers for MP3). Then suppose you have a low-latency codec (like Opus) with a 15ms frame size. In both cases, your network latency is going to be the same (let's say 100ms).
You want to stream audio over this connection. Kudos to the folks working on this. We were all rooting for ogg/vorbis/xiph, but they had some lessons to learn. Positives that I see for Opus:. libopus is available now. it has an integer-only compile flag. it's BSD licensed.
patent grants from big industry players. doxygen API docs. big open source projects already support it.
orchestrated PRstill could use some love:. apparently it's slashdot.org but that's not bragged about (and many would suspect otherwise).
some of the documentation is just a link to slide decks from conferences. there is test code, but I didn't see sample code explicitly. Yeah, you can grab ffmpeg source or whatever, but purposeful sample code is written to be as explanatory as possible.
Maybe it's in the tarball, but if it is, say so on the download page.Still, an order of magnitude better than the last attempt at gaining industry acceptance of free codecs. This one might just work out! To me the biggest difference is that Vorbis was competing head on with a strongly entrenched codec (MP3) and it's official successor (AAC).
Opus on the other-hand fills niche in the audio encoding world that doesn't have an established winner; that is high-quality low-latency codecs. This area has largely been driven by cellphone market, and has focused on encoding voice signals at toll-quality, that is as good as an analog long-distance signal (8kHz mono). There really hasn't been much focus on creating a low-latency codec that can encode full-band (music signals), and Opus does that incredibly well. It also sounds much better encoding speech at the bitrates that are used for VoIP (rather than the lower ones used by cellphones).The internet community has never really been happy with the performance of ITU specified codecs that have been primarily used for SIP and other VoIP applications in the past, and there is no good reason from them not to support Opus. The patent grants are there, the vender support is there, and there is no real competitor codec worth mentioning. I'm convinced this will make much deeper inroads than Vorbis did. 'Opus covers basically the entire audio-coding application space'Maybe I didn't look hard enough but I didn't see anything about how well it handles getting some of it's data corrupted.
I only see comparisons of how it works at different bitrates. This is important for radio applications as there will always be interference and some percentage of the received bits will be wrong. That is why for example we don't see Amateur Radio operators using Speex. If this truly covers everything then we don't need codec2 codec2.org but from what I see it just sounds like a new ogg vorbis which is useful through a wider range of bitrates. Last I checked, APT-X was not lossless (you can't be lossless and guarantee a compression ratio). Also, the only time I saw a comparison between APT-X and Opus, Opus was actually winning hands down.
APT-X claims compression ratios of 4:1 (so 384 kb/s for 48 kHz stereo). Opus is already transparent long before that. I've never had anyone telling us he could hear any kind of artefact whatsoever above 200 kb/s VBR). Usually, 128 kb/s (12:1) is transparent for the vast majority of content and listeners (yes, th.
. audiophile: a person with love for, affinity towards or obsession with high-quality playback of sound and music.is a forum for discussion of the pursuit of quality audio reproduction of all forms, budgets, and sizes. Our primary goal is insightful discussion of equipment, sources, music, and audio concepts. Rules. Be most excellent towards your fellow redditors. And by 'be most excellent' we mean no personal attacks, threats, bullying, trolling, baiting, flaming, hate speech, racism, sexism, or other behavior that makes humanity look like scum. Ask for product opinions and purchase advice in the.
This includes general questions or comparisons about gear and peripherals regardless of intent to purchase. Post questions about tech support and general help in the. Low effort questions also go here. Image posts must be accompanied by impressions or a review that adds value to the post.
The impressions or review do not need to be exhaustive, but they should strive to explain how you feel about the product(s) and why you feel that way. No pictures of unopened boxes!. We do not allow: affiliate links, links to affiliate farms, pirated content, NSFW/NSFL content, market research, surveys, sweepstakes, giveaways, spam or self promotion. No selling or buying. Please use. No headphones and portable audio related content.
Please useAdditionally,. We may further remove posts that are deemed off-topic, or better suited to other subreddits. What we want to see.
Content that facilitates about audio quality. New hardware. Gear.
Reviews of audiophile. Discussions on. of your setup. Or someone else's if it gives audiophiles 'The fizz'Post not appearing?If you made a post to that is in accordance with our rules as listed above, and it doesn't seem to appear on the front page, please. Related subreddits.
Break AUTOCAD 2004 activation. This crack will work with 2 phases. First of all open notepad and copy the code below and paste in notepad. Save the file as 'LICPATH.LIC'. File is in the '.LIC' format. On desktop or anywhere. Autocad 2004 authorization code, AutoCAD 2010, AutoCAD 2008, Authorization Wizard 2.1. May 11, 2018 I recently reinstalled windows and put Autocad 2004 back on there but it asks for an authorization code. It did not when I installed from disk. Just to be clear I am not asking for the codes, I just need an explanation. Authorization code autocad 2004 free. Dec 16, 2011 I need an authorization code for Autocad 2004. Product: AutoCAD 2004 Serial number: 11 Request code: 0525 2149 2686 4964 5032 pls get me the authorization code as soon as possible its very urgent. To get the activation code you should BUY Autocad. If you want a free product, look for FREE PROGECAT 2009. AutoCAD Compatible.
Headphones and portable Audio. Music for audiophiles.
Pro audio/engineering. Do-It-Yourself Audio - also and. for restricted budget hifi. DIY Headphone mods. Headphone pictures. Repair help for audio gear.
Trade used AV gear. AV pictures. Car Audio and Video. Vintage audio gear. Turntables & RecordsModerators may at their discretion remove content that fits better in one of the above subreddits.Subreddit's theme is.
So I made a comparison of FLAC, AAC (Apple) and OGG (Vorbis) to see which one of the better lossy formats is closer to lossless in quality.In this comparison I used 44.1kHz/16bit FLAC tracks that I converted into 320kbps AAC and 320kbps OGG files. Later these files have been converted back to FLAC and compared to the original FLAC file.
The analyses were done in Spectro.I analysed five songs from different genres, here are the results:.In summary, 320kbps AAC is very damn close to lossless, however, OGG isn't that effective. Sometimes there's a significant drop in higher frequencies and the difference in the spectrogram is noticeable. Overall it's also close to lossless and I'm sure the difference between these 3 formats is inaudible to most (if not all) of you. Viewing spectrograms isn't a foolproof method of determining audio quality. Sure, with the appropriate settings and time scale you get an overview of what frequenices are missing in a lossy version compared to lossless, but this completely disregards the fundamental factor that makes lossy (aka perceptual) encoders work - i.e.
The psycho-acoustic models used that are designed to fool human ears and brains - i.e. The things we actually hear with, and not our eyes. You wouldn't judge the quality of a photograph by listening to it.Spectrograms certainly have their uses, but making judgements as to the quality of lossy codecs based on what you see, really isn't one of them. All lossy codecs have some killer samples that sound bad even with a fine looking spectrogram, and most lossy encoders can achieve transparency with most music at bitrates well below 320kbps. These 2 facts are not mutually exclusive - the source material always matters.