A better AAC encoder: FDK-AAC

It really works, thanks!

Quote from poo

It would be nice if you added ability to use Apple CoreAudio AAC. I'm sure its possible to do since OBS does this too (when its installed on system).

Any updates on this matter and if it's possible to add support for it? Sure would help to streamline things a little bit, for me at least.

Simple question: the initial download "libfdk-aac.zip"... is the latest version of FDK AAC Encoder??

VBR in HE-AAC is possible now??

I noticed the audio streaming client butt also offers libfdk-aac-2.dll for its users, it's 1.3 MB and supports VBR, HE-AACv1 (AAC+) and HE-AACv2 (eAAC+) with Fraunhofer's qualitative tuning, VBR & CBR and LC-AAC.

Quote from spreeni

I did some tests with a sequence of white noise. I made a screenshot of the spectralview in Audition. The results are attached as PNG-pictures:

01-uncompreessed.png: Uncompressed - this is what it shoul look like

02-AAC-ffmpg_CBR_320.png: The native FFMPG AAC Encoder at 320kbit (48Khz stereo) - look at these "glitches"

03-AAC-ffmpg_CBR_384.png: Same settings but slightly higher bitrate (384kbit) - noticeably better quality

04-fdkAAC_CBR_320.png: fdk AAC Encoder at 320kbit, very good quality, but frequencies above 17Khz are cut

05-AAC_x264Pro-plug-in: AAC Encoder of the x264Pro plug-in. I don't know what encoder is used here but it looks outstanding

So my result is: For best AAC quality in voukoder use the native FFMPG AAC encoder at higher bitrates (min 384) or use the the fdk encoder and live without the 17.000+ frequencies. After some rock concerts these frequencies are blows away, anyway )

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Thank you for this info! I used to compile FFMPEG with fdk aac built in and I was considering doing this again until I saw your post here.

Which profile are you using for 03-AAC-ffmpg_CBR_384? Low complexity?

Is this still one of the best audio encoders you've found that works with Voukoder?

I just use QAAC's cvbr mode for best compression and quality possible among 99.9% of audio encoders:

qaac64.exe --cvbr 320 -b 16 -r 44100 --threading -o "output.aac" "input.wav"

Sometimes I have to lower the cbvr bitrate (e.g., --cvbr 256 for 256Kbps) to make the export compatible with some audio sharing platforms.

HE-AAC does offer more compression, in QAAC it allows a maximum of 80Kbps, which butchers the audio quailty, though

Seems this thread gets revived every now and then so here is a question from me

Regarding a previous post:
But there is an alternative / workaround:

• Simply extract the libfdk-aac-2.dll file from the attached zip archive to your C:\windows\system32\ directory.

Now everything is always about fdk-aac or Qaac.

Many years back i have held a true double blind test focussed on music 14 X random A - B test, with me (audio engineer) and 2 other audio guys comparing the different aac encoder as wel as mp3, vorbis and OGG.

keeping it short the FHG-AAC was the only aac that we could not distinguish from lossless (highest vbr preset)

Is there an easy way to import or start using this encoder in voukoder? I know it can't be included licence wise and stuff, but for example in foobar i can just select this as my default aac encoder.

The version of the encoder i have is a combination of the fhgaacenc.exe console shell and the enc_fhgaac plugin for Winamp.

- work in CBR (8-576 kbps) and VBR (6 quality presets)
- support for encoding with STDIN
- supported audio: 8/16/24 bit, 22050-96000 Hz, up to 8 channels
- support SBR and PS (HE-AACv2)

Probably ist that simple of just copy paste files to windows or a voukoder folder regarding this?

Could we get an audio cutoff setting like in FFmpeg? That would solve the issue with the audio above 17KHz getting cut off. By default libfdk_aac applies a low pass filter but the ffmpeg command paramete "-cutoff 20000" sets the low pass filter to 20KHz (which is the max, if set higher I will error and tell you 20KHz is max.)

One screenshot is running ./ffmpeg -i '.\Katy Perry - Teenage Dream - 04 - Firework.flac' -c:v copy -c:a libfdk_aac -vbr 4 KP-regular-vbr-4.m4a and the other is running ./ffmpeg -i '.\Katy Perry - Teenage Dream - 04 - Firework.flac' -c:v copy -c:a libfdk_aac -vbr 4 -cutoff 20000 KP-20k-vbr-4.m4a. In the screenshots, you can clearly see one has much lower levels above about 17KHz and that is the first command. The other has normal sound levels above 17KHz and that is with -cutoff 20000. I'll try and test these audio files with Voukoder and see if the cutoff applies or if it's a weird FFmpeg thing. I will get back to you in a second comment when I check.

Voukoder has the same cutoff applied by default. This could be useful: https://trac.ffmpeg.org/wiki/Encode/AAC#fdk_aac

So it's not just FFmpeg, it's the FDK AAC encoder.

Chapter 3.1 which is on page 23 of the FDK AAC docs explains something about the bandwidth and the API for it: https://github.com/mstorsjo/fdk-a…/aacEncoder.pdf

On this FFmpeg documentation page it lists cutoff as an option:

ffmpeg Documentation

Quote from Vouk

I'm investigating this again.

Edit: Both AAC codecs get the option to override the cutoff frequency in the next release.

Woohoo!

I ask you, why use .aac if in HEVC H265 it is not a format accepted by 4K UHD (unlike .ac3)?

Thanks

Quote from Promix

I ask you, why use .aac if in HEVC H265 it is not a format accepted by 4K UHD (unlike .ac3)?

Thanks

Rephrase that please, what you said doesn't make sense.

Quote from Promix

I ask you, why use .aac if in HEVC H265 it is not a format accepted by 4K UHD (unlike .ac3)?

Thanks

Putting aside video, stream files are a list of channels and bandwidths. If I listen to radio on my phone, I don't have 7.1 speakers in the first place.