Why xHE-AAC is being embraced at Meta(engineering.fb.com) |
Why xHE-AAC is being embraced at Meta(engineering.fb.com) |
Also, I wish they had backed opus…
or, more likely, lower bandwidth at similar quality.
Hmm... You could detect the output type (built-in speakers, headphones, bluetooth) and change the audio quality accordingly.
The only good customer is one that willingly opts into all ads I guess.
There's indeed some truth to that as well in the sense that the customer is the one who wants the ad to be displayed in the first place and pays for it. So it's a form of opt in as well.
That being said, the user who sees the ads is not the customer but the product. And yes, the best sellable product (i.e. user) for Meta (Facebook, Instagram etc), Alphabet (Google etc) and the like is the one that willingly opts into all ads.
Except... I wanted to stream music across the network to another computer with the lowest possible latency. I didn't care about the extra quality you get from lossless (opus sounded fine to me at 96kbps), but chose flac because it has almost no latency. Opus (and for that matter, all other codecs) added noticeable latency.
I used Opus for a VOIP app and I was really impressed with it. It's all I would ever use for Internet audio.
[1] https://opus-codec.org/docs/opus_api-1.3.1/group__opus__enco...
There is also this product by the Nullsoft guys, no idea if it's any good though: http://cockos.com/ninjam/
I think Opus also has loudness management, but they seem fairly specific about the one that xHE-AAC provide : > Instead of burning in a specific target level and dynamic range compression (DRC) profile during encoding, xHE-AAC allows us to leave the original audio characteristics untouched and delegate loudness management processing to the client via loudness metadata, for the optimal audio experience based on context.
So maybe the xHE-AAC does provide a tangible benefit in this department.
Less bandwidth. Some users pay for bandwidth. Or have a monthly cap. So less bandwidth is a monetary advantage for users. It also may decrease time to play.
Similar, but less ctitical, for storage.
One rule of thumb when switching codecs is "don't decrease quality". I buy that Meta did that. Which would result in a slight improvement in quality (of course sold as big snd noticable) while getting storage and bandwidth down (or allow for better quality in the bandwidth adaptive case).
That saud. Yes. OPUS! Although other aites got quite some bad press for low bitrate opus.
For example, for most modern formats software-only encoding is very slow, and you really want to have hardware support for it [1]. But to add hardware support to increasingly complex and varied specs is not a decision you make lightly.
[1] Google is forcing companies to support its codecs in hardware on pain of removing Youtube e.g. https://arstechnica.com/gadgets/2021/04/roku-vs-google-part-...
So it is supported on the web.
>This was probably done to reduce storage and bandwidth, not to improve audio quality.
Packing more information into the same amount of data is the literal definition of better quality.
>Phone speakers aren’t good enough to notice.
Maybe not, but phones can be connected to headphones and speakers which can. Especially if they connect via 3.5mm audio jack.
Topic: A Session In The Abyss: xHE-AAC vs OPUS at 12, 24 and 32 kbps (voice & music) (Read 16655 times) https://hydrogenaud.io/index.php/topic,120997.0.html
xHE-AAC vs HE-AAC v2 Audio Quality Comparison: War of the Low-Bitrate Codecs https://www.youtube.com/watch?v=74SsKOUHgvo
These days even YouTube videos get transcoded to 128 kbps Opus, which is astonishing. Provided that the source is high enough quality, you can listen to transparent audio on the lowest-tier streaming platform.
[1] https://wiki.hydrogenaud.io/index.php?title=Hydrogenaudio_Li...
What I found a few years ago was that explicitly lowpassing beforehand to match the other encoders (around 4-8kHz) gets rid of the buzzing. Apparently opus' threshold for automatically doing this is a lot lower than the other encoders; it keeps higher frequencies at the cost of more artifacts. I concluded that decimating or setting the encoder to 8kHz/"wideband" or lower was an improvement, with a similar resulting quality to AAC, but I still slightly preferred Fraunhofer at 12-28kbps.
By default, 12 and 24kbps are particularly bad; at 16k the quality goes up from going mono-only, and by 8k bandpassing is automatic.
to be fair that tester indicated only voice being ok, and everything else just bearable under condition you are consciously working under severe bw constrain "EDGE poor network is still enough to stream music without issue (in a tunnel, on mountain…)"
The main features focused on in this article are loudness management and adaptive bitrate audio. From what I can tell Opus supports neither of those things.
Loudness management isn't really the purview of the decoder/compressed format itself; the article describes using MPEG-D DRC which is, at least in principle, independent of the choice of xHE-AAC vs. Opus. To be clear I have no idea how companies do dynamic range compression in practice; in the Web setting, maybe via stuff like https://developer.mozilla.org/en-US/docs/Web/API/DynamicsCom... etc.?
I'm not convinced the loudness metadata makes much difference, especially for the entity doing the encoding. Can it do compression at playback time too? That's sort of implied but I'm not sure.
[edited: punctuation]
<img title="Audio stock hero">Edit: I'll elaborate here. I went all sciencey and wrote a script to test all combinations of a range of encoder settings that included ranges of comp, max delay, frame size, bitrate etc. Multiplied out, that was a lot of different combinations. I ran the script (which took ages because I had to record the startup latency for each one and capture audio on the remote end, logging ffmpeg output etc etc). Sometimes a particular setting was so bad across 5 tests (with different values for other settings) that I'd scrap that whole batch to save time. The ranges of configurations that were promising were used "in real life" to play music in our living room. I narrowed them down to a few that were "usable", but I asked myself "if I bought this thing as a product, would I send it back?". The answer was "yes".
FWIW, the sender was a Macbook Pro 2015 and the receiver was a Raspberry pi 4B
TL;DR At or below 5ms it had reliability issues (drop-outs) that would occur randomly but frequently enough to ruin music listening experience (like, even a 1s drop per song is completely unacceptable). But it also seemed to stretch audio because the delay would increase over time. Even when I got a high quality (zero drop-out steam) with acceptable latency, the latency would slowly increase over time. So about 1 hour into listening, of I paused the source to answer a call, the decoder end would continue playing for 30+ seconds.
I realised that over a LAN, streaming raw audio technically works, with almost zero latency, so I knew that it was the encode/decode that introduced the latency. That's when I switched over to flac, and noticed an immediate improvement in initial latency, but also stopped having the increasing delay over time.
I'm not aware of any hardware acceleration.
Yes, it has dynamic range control too.
Overall I just found flac to be rock solid and lower latency without much tweak
https://developer.android.com/guide/topics/media/media-forma...
https://developer.apple.com/documentation/http_live_streamin...
Does anyone know why it isn't possible to add codecs to the browser?
Mozilla famously relied on Cisco to gain h.264 playback, because the patent included a certain maximum total price you needed to pay after exceeding a certain amount of devices. Cisco exceeded that amount, so any additional devices they supported were practically free, and they released a free h.264 decoder plugin for Firefox. The pricing loophole has been removed out of h.265 so a Mozilla decoder would cost them a lot of money.
In theory you could add codecs to your browser just fine, but Mozilla has already indicated it's not planning on adding h.265 to their browser, focusing on AV1 support instead.
Perhaps Mozilla would welcome patches to allow forwarding h.265 streams to the OS for decoding, like Chrome does, but they're not going to put the effort in themselves. With hardware AV1 decoding finally on the rise, I don't think that's a bad decision necessarily, companies and websites opting for the patent ridden format over the open format knew what they were doing.
No, they didn't. OpenH264 only supports Constrained Baseline profile, and is only useful for WebRTC. It's used system libraries/frameworks for H.264 video playback since long before OpenH264 came around.
Firefox is open source, however, so there's no reason why a fork couldn't add that functionality. I personally wouldn't put any development effort into it myself and Mozilla don't seem interested, but for people who feel strongly about this there are always options.