Help with encoding
 

Hi i want to know what is the difference with (MP4/accPlus HE-ACC Hight Bitrate Encoder v1.2) VS (accPlus HE-ACC Hight Bitrate Encoder v1.2 MPEG-4) ?

Please Let me know thanks.

One is aacPlus inside an MP4 container (.m4a extension), the other is raw aacPlus with an .aac extension. Other than that, they're pretty much the same.

Do you know what of the two encoding have the best quality in aacPlus?

I think there is no quality difference. The only difference is container and extension (.m4a and .aac).

It is true that the best encode quality for music is aacPlus?

magnavex: depends on what you want to use it for.

lossless compression or no compression at all will provide the best quality files, so they should sound EXACTLY bit for bit like the original source.

examples of lossless compression
flac(*.flac), monkeys audio(*.ape), wavpack(*.wv), shorten(*.shn)

examples of no compression
PCM, LPCM.... usually in *.wav containers(*.wav)

if your going to use it in a portable at some time then mp3 with --alt preset-standard will be the best and have the best compatability.(*.mp3)

ogg vorbis is also VERY VERY good audio compression for almost any purpose provided that you have an mp3 player that supports it... like an iriver.(*.ogg)

aac(mpeg 2 part 7) is just another mpeg 2 format which is a bit more efficient than mp3(aka mpeg 1 layer 3). i tend not to use it because it has less compatibility with portables, and i can't tell the difference in sound between mp3 and aac.(*.aac, *.m4a)

aacplus(mpeg 4 part 3) is a mpeg 4 variation of aac which was originally made for internet streaming, it is very efficient at low bitrates.


I personally still choose mp3 or ogg vorbis for my ripping purposes. ogg gets used a bit more these days as it sounds better than the other formats, and its free to use without any licencing stuff.

If you have to choose whether to use the MP4 container, or not, always use it since it is the more standard way to handle AAC/aacPlus files (apples iTunes,...).

AAC or 'Advanced Audio Coding' forms part of the latest specifications from the MPEG committee, and is their official successor to the popular MP3 format. As with MP3, the AAC format is an international standard, and is backed by several big-name companies, including Dolby, Sony and Nokia.

With the 13 years that had passed since the creation of the MP3 format, many improvements had been realised leading to a seemingly complex specification with several flavours of AAC available. To potentially add to the confusion, AAC is usually wrapped inside an MP4 container to provide tagging, seeking and possibly other benefits?? For this reason, AAC can also be referred to as MP4 audio..

There are several AAC encoders to choose from, coming from large names such as Apple (iTunes) , Real Networks and Nero AG (Creators of Nero Burning Rom), or the open source FAAC which is analogous to the LAME encoder. AAC is supported on some hardware players, most notably the Apple iPod and some cell phones, and is available in Apple's online store.

In terms of quality, the AAC format is on par with Ogg Vorbis, LAME MP3, WMA Pro and other modern codecs, and with added SBR coding (HE AAC) it can provide quite high quality at low bitrates.

Recent developments have led to AACplus which is able to give subjectively good results at low bitrates. The website http://www.tuner2.com has several Internet radio stations which are sending out streams at low rates - such as 40 kbps - and some of these are surprisingly good considering the bit rates used.

Pros

* An international standard approved by the ISO
* Flexible: supports several sampling rates (8000-96000 Hz), bit depths, and multichannel (up to 48 channels)
* Several implementations, including free and high quality ones (iTunes or Nero Digital)
* Reaches transparency in most samples and for most users at around 150kbps
* Part of MPEG-4 specs
* Anyone can create its own implementation (specifications and demo sources available)
* Some portable players support it (Philips Expanium, Apple iPod, cell phones from Nokia, Sony Jukebox)

Cons

* Problem cases that trip out all transform codecs
* Heavily patented
* Increased complexity
* AAC comes in different "flavors" (object types: AAC LC, AAC HE, AAC PS etc.). Many (especially portable) players only support LC (at the moment) so you can have files that are valid but your player won't play them.

Ogg Vorbis (www.vorbis.com) is a fully open, non-proprietary, patent-free (subject to speculation), and royalty-free, general-purpose compressed audio format for mid to high quality (8khz-48.0kHz, 16+ bit, multichannel) audio and music at fixed and variable bitrates from 16 to >256 kbps/channel. This places vorbis in the same competitive class as audio representations such as MPEG-4 (AAC), and similar to, but higher performance than MP3, TwinVQ (VQF), WMA and PAC. Vorbis is the first of a planned family of Ogg multimedia coding formats being developed as part of Xiph.org's ogg multimedia project.

Informal listening test suggests Vorbis to be comparable to MPEG-4 AAC at most bitrates and MPC at 128 kbps. Transparency is generally reached at about 150-170 kbps (-q 5) (with some exceptions). The encoder is reasonably young and unoptimized, so further improvements can always be expected.

Unfortunately, Xiph.org has failed to improve Vorbis at a steady rate since its initial 1.0 release in July 2002 (due to other developement projects and time constraints). Since then development has been led by other coders such as Garf and Aoyumi. Aoyumi's aoTuV series of encoders was incorporated into the September 2004 release of 1.1, which brought about the first quality improvements across the board for 2 years. Currently Aoyumi is working on aoTuV Beta 4 and future releases. The latest version is aoTuV Beta 4.51, released in December 2005. Unfortunately, the improvements of aoTuV Beta 4 line has not been incorporated yet into the 'official' Vorbis line.

Vorbis has had success with many recent video game titles employing Vorbis as opposed to MP3 (with Epic Games' Unreal Tournament 2003 and Unreal Tournament 2004, the PC port of Microsoft's Halo and Uru being notable examples). Ogg Vorbis is also an official part of the OpenAL API extension library, used in many popular computer games. On April 10, 2006, RAD Game Tools integrated Ogg Vorbis support to their Miles Sound System (MSS), which has been used in over 3,200 games worldwide. This ensures that future games utilizing MSS will have the capability to play Ogg Vorbis files. Check out xiph wiki for a full list of games confirmed to use Ogg Vorbis.

Before encoding files using Ogg Vorbis, check out the Recommended Ogg Vorbis to determine what encoder to use and what settings are recommended by Hydrogenaudio.

Pros

* Ogg Vorbis specification is in the public domain; it is free for commercial or noncommercial use, under both (LGPL and BSD liscenes)
* Good all-round performance (>48 kbps - a leading codec at 128 kbps)
* Well written specs
* Supported by most portable DAPs
* Suitable for internet-streaming (via Icecast and other methods)
* Fully gapless playback
* High potential for further tuning
* Structured to allow the design for a hybrid filterbank

Cons

* Limited official development (third-party developement is always encouraged)
* Current implementations are more computationally intensive to decode than MP3
* Multichannel input mappings for 5.1, Ambisonic-B, and other configs have no channel coupling and aren't tuned (expect sub-optimal results until code is improved)

MP3 MPEG 1 Layer 3

The MP3 algorithm development started in 1987, with a joint cooperation of Fraunhofer IIS-A and the University of Erlangen. It is standardized as ISO-MPEG Audio Layer-3 (IS 11172-3 and IS 13818-3).

It soon became the de facto standard for lossy audio encoding, due to the high compression rates (1/11 of the original size, still retaining considerable quality), the high availability of decoders and the low CPU requirements for playback. (486 DX2-100 is enough for real-time decoding)

It supports multichannel files (Although there's no implementation yet), sampling rates from 16kHz to 24kHz (MPEG2 Layer 3) and 32kHz to 48kHz (MPEG1 Layer 3)

Formal and informal listening tests have shown that MP3 at the 160-224 kbps range provide encoded results undistinguishable from the original materials in most of the cases.

Pros

* Widespread acceptance, support in nearly all hardware audio players and devices
* An ISO standard, part of MPEG specs
* Fast decoding, lower complexity than AAC or Vorbis
* Anyone can create their own implementation (Specs and demo sources available)
* Relaxed licensing schedule

Cons

* Lower performance / efficiency than modern codecs.
* Problem cases that trip out all transform codecs.
* Sometimes, maximum bitrate (320kbps) isn't enough.
* No multichannel implementations.
* Unusable for high definition audio (sampling rates higher than 48kHz).

is it me, or did he JUST answered his own question with that last bit? You found your comparison there Magnavex
good luck in your journey

Yeap i found the answer in this site http://wiki.hydrogenaudio.org/index.php?title=Main_Page

Thanks to all.

encoding settings to use?
 

Hi, I just upgraded to 5.3

I seem to have lost all my settings and have new defaults for encoding.

Up to then I'd been using LAME VBR 128/360, which served me fine.

The new default seems to be

mp4/lc-aac encoder v1.21

Okay, I know how to encode through LAME, but can mp3 players (such as Creative) have the capability to play lossless formats?

How does file size of mp4 compare to mp3? Is lossless encoding still mainly for archiving instead of playing?

The defaults for the mp4/lc-aac is 128 bitrate, stereo. Is there any reason to move that higher?

Aside from ogg, is there any reason to believe that mp4 encoding wouldn't play on linux laptops (assuming you download the right libraries)

rj

As far as I know Creative does not support lossless formats. Other players may support lossless formats. The only other player I have is WMP and it does at least support flac.

That really depends on the quality of equipment you will playing your music with and how acute your ears are. The only true way to determin is by ABX testing. Check out the Hydrogen Audio forum if you want more info. Generally speaking, 128KB CBR is not transparent and on decent equipment, in a quite environment, you likely will hear a difference. Personally I would suggest encoding to mp3 or ogg vorbis. Both of these encode with VBR which is much more efficient than CBR. You can play around with quality settings until you find the setting that is transparent to you. The Wiki page produced by Hydrogen Audio can guide you in your quality selction. (I would provide a link but am too lazy to search for it.)
File size is directly linked to the quality selection chosen. Mp3 and M4a have comparable file size at comparable qualities.

If I had the hd space all of my music would be lossless. As it is I store my important stuff on disc using flac. By encoding to a lossless format you have the capability to transcode to any format of any quality for specific purposes. But if you rip to a lossy format you may suffer significant loss in quality if you transcode those files.