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12th December 2016, 17:55 | #1 | Link |
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Encoding witth QAAC like CT AAC
Hello there,
I've been using Coding Technologies AAC from a long time, with WinAmp old dll, using BeLight and BeSweet. The parameters I used the most often are HE-AAC 2.0 64 kbps and LC-AAC 5.1 192 or 256 kbps, sometimes HE-AAC v2 with Parametric Stereo 32 kbps. I'm very satisfied with the result since i'm just looking for medium quality audio for 700 MB movie rips. I recently discovered QAAC, and the fact that it can be directly encoded through MeGui, without using BeSweet. So my question is really simple, what parameters should I use in MeGui to have the same (or better ?!) results at the same bitrate using the QAAC encoder and MeGUI? Thanks for your help! |
12th December 2016, 21:56 | #2 | Link |
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You can use eac3to, BeHappy or MeGUI to encode to AAC with QAAC.
If you want fix a average bitrate you can use the 'Constrained VBR' mode, always better than ABR or CBR . After you can select the Profile desired. And, for movies, I recommend check the 'No Delay' option.
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12th December 2016, 22:21 | #3 | Link |
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Thanks for your help!
So, for example, for my first stereo HE-AAC 64 kbps setting, should it be like that: And for 5.1 LC-AAC 192 kbps setting: What if the source is a 7.1 DTS, will the LC-AAC be 7.1 to, or is there an option to keep 6ch instead of 8 ? Or does the AAC only allow 6ch max? I don't find an equivalent of Parametric Stereo for low bitrates (e.g. 32 kbps)... And last question, sometimes, for a 6ch track, I used to check the "Mode Boost: LigH" option in BeLight, to boost the volume for people who listen to the track in on only stereo system, is there such an option for qAAC in MeGUI, or in BeHappy ? |
13th December 2016, 15:17 | #4 | Link |
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My first answer was only related with QAAC encoder.
QAAC don't support HEv2 (Parametric Stereo), is very low quality not recommended for track movies. EDIT: seems change automatically for low bitrates like herbert show in second next post EDIT 2: I was right QAAC don't support HEv2 Other question is how decode, downmix or improve the audio before encode: - Preferred decoder: Use always 'LWLibavAudioSource' for any input (DTS-HD, TrueHD, DTS, E-AC3, AC3, AAC, ...) - Output Channels: Use 'Keep Original Channels' or 'Downmix multichannels to stereo' at your choice. MeGUI don't offer 7.1 to 5.1, and encode to a 7.1 AAC is possible but not recommended. Maybe you need BeHappy to do 7.1 (DTS-HD or TrueHD) -> AAC 5.1 - Apply Dynamic Range Compresión: recommended if 'Downmix multichannels to stereo' but only work with AC3 sources. - Normalize Peaks to: 100% recommended if 'Downmix multichannels to stereo' (equivalent to BeLight OTA) The BeLight (BeSweet) Boost filters aren't supported here, we can do some similar effects with AviSynth plugins, but is not easy to implement.
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BeHappy, AviSynth audio transcoder. Last edited by tebasuna51; 15th December 2016 at 17:17. Reason: add info |
14th December 2016, 00:04 | #5 | Link |
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Thanks for the details. I only use HE-AAC v2 (Parametric Stereo) for commentary tracks, which don't need high or medium quality audio, it's just to save half the size of a track! I'll keep using BeLight for this, and for 6ch tracks (to be able to boost a little), too, thanks again for your help
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14th December 2016, 01:25 | #6 | Link |
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Apple's encoder switches to HE-AAC v2 (PS) automatically for 32kbps or lower.
Code:
ffmpeg -i test.mp3 -f wav - | qaac64 --cvbr 64 -o lc.mp4 - AAC-LC Encoder, CVBR 64kbps, Quality 96 Format profile : LC ffmpeg -i test.mp3 -f wav - | qaac64 --he --cvbr 64 -o he.mp4 - AAC-HE Encoder, CVBR 64kbps, Quality 96 Format profile : HE-AAC ffmpeg -i test.mp3 -f wav - | qaac64 --he --cvbr 32 -o heps.mp4 - AAC-HE Encoder, CVBR 32kbps, Quality 96 Format profile : HE-AACv2 |
14th December 2016, 15:10 | #10 | Link |
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Hmmm, interesting.
This was my test file: http://www.linguistik.uzh.ch/static/...en-2016-11.mp3 MediaInfo tab in MPC-HC lists this as HE-AAC v2 when encoded at or below 32kbps by QAAC. |
14th December 2016, 21:56 | #12 | Link |
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Using qaac v2.61 the low bitrate supported is 32 Kb/s and the output is HE, not HEv2, with 2 standard channels +SBR
Using the mp3 uploaded or any input than I test. Code:
angesprochen-2016-11.mp3_.m4a AAC-HE Encoder, CVBR 32kbps, Quality 96 [9.0%] 24:23.222/4:30:31.923 (23.0x), ETA 10:42.577 64528128/715827840 samples processed in 1:03.679 Overall bitrate: 31.9982kbps Optimizing...done
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14th December 2016, 22:47 | #15 | Link |
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Quite possible, I'm using MediaInfo as shipped as part of MPC-HC 1.7.10.264.
ffmpeg version N-82225-gb4e9252, qaac 2.58, CoreAudioToolbox 7.10.7.0 I uploaded test encodes here: https://mega.nz/#!pYgTDQSI!6G1MQCAAd...EZO4jRg76326L0 testffmpeg.mp4 = ffmpeg -i angesprochen-2016-11.mp3 -f wav - | qaac64 --he --cvbr 32 -o testffmpeg.mp4 - test.mp4 = qaac64 --he --cvbr 32 -o test.mp4 angesprochen-2016-11.mp3 The file generated by ffmpeg | qaac is identified as HE-AAC v2 by MediaInfo. The plain qaac encode is identified as HE-AAC. Is there any reliable way of identifying the actual version of HE-AAC employed here? Edit: Changed host for files Last edited by herbert; 14th December 2016 at 22:57. |
15th December 2016, 01:24 | #16 | Link |
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I have my own tool to check .m4a
This is a true HE v2 encoded with fdkaac: Code:
File ........: D:\tmp\angesprochen-2016-11.m4a Size ........: 5993053 bytes ---------------------------------------------- Header Info Atom ID .....: ftyp Audio format : AAC-HE v2 (LC+SBR+PS) Channels ....: 2 SampleRate ..: 44100 Hz. Total Frames : 31511 Total Samples: 32267264 Data Length .: 5853472 bytes Duration trak: 1463.368 sec., (0h. 24m. 23.368s.) BitRate .....: 32 Kb/s Code:
File ........: D:\tmp\test.mp4 [testffmpeg.mp4] Size ........: 5997310 bytes [5999452 bytes] ---------------------------------------------- Header Info Atom ID .....: ftyp Audio format : AAC-HE (LC+SBR) Channels ....: 2 SampleRate ..: 44100 Hz. Total Frames : 31510 Total Samples: 32266240 Data Length .: 5857602 bytes [5859844 bytes] Duration trak: 1463.322 sec., (0h. 24m. 23.322s.) BitRate .....: 32 Kb/s
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BeHappy, AviSynth audio transcoder. Last edited by tebasuna51; 15th December 2016 at 02:01. |
15th December 2016, 19:22 | #17 | Link | |||
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Quote:
Quote:
First, never use the DspGuru mode, it is broken cutting and inversing peaks. The LigH and Tera mode work amplifying low volume of all channels, and let high volume without changes. If the problem is low dialog volume in 5.1 I think than is better amplify only the center channel Also BeLight work only with AC3 input (for 5.1). You can use a script like this: Quote:
To accept any source even DTS-HD or TrueHD 6.1 or 7.1 than downmix to 5.1. If you remove the '#' in: cen = Getchannel(a, 3).Amplify(2).SoftClipperFromAudX(0.0) you can amplify only the center channel at your desired level.
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BeHappy, AviSynth audio transcoder. Last edited by tebasuna51; 16th December 2016 at 10:57. |
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16th December 2016, 23:59 | #19 | Link | |
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Quote:
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17th December 2016, 23:30 | #20 | Link |
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OK my first try is not really succesfull ^^
Doing a simple DTS -> HE-AAC conversion with these settings getting this error Did I miss something? BTW I tried the same file 2 days ago but I wasn't home when it finished. I closed MeGUI without seeing that there were errors and the file didn't encode at all, I got a .dts.lwi file and an empty aac file with the same filename but I couldn't see the log... |
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