I recently started re-encoding some of my high bit rate videos to save space using Handbrake. The problem is that I may have set the audio quality a bit too low. I went from 6 channel FLAC to 6 channel AAC at 384 kbps. I regret doing this, but now I have no way to re-encode only the audio portion using Handbrake. I'm sure I'm not the first person to make this mistake and I'm hoping I could get some suggestions on how to proceed.
A better choice would have been to go with AAC at 1536 kbps or maybe even DTS at that bit rate? I was thinking of using Avidemux to re-encode the audio only. I remember Avidemux having a "null" option for video, which should allow audio only re-encoding. Would that be a good way of doing it? Are there any free utilities that can re-encode to DTS?
I'm no audiophile, but I want to keep my audio high quality without paying for expensive software to do it. Another thing I'm worried about is the audio being out of sync if I use a separate program to encode it and then mux it with the video. I normally use mkvmerge for muxing and it's been pretty reliable.
Any suggestions regarding this would be greatly appreciated.
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XMedia Recode has the option to "Copy" (instead of Converting) the video and encoding the audio only (and remuxing them) - I checked and it will convert to DTS at various rates . It will also allow adding in a set of videos, highlighting them and setting common settings to them all/batch them .
Last edited by Sartori; 5th Jan 2018 at 12:02.Llamas are for life , not just for christmas
just correction, if you want keep video in avidemux, leave it on copy mode (default).
But only audio format supported by avidemux are 2 codecs of AAC,2 codecs of AC3, 2 codecs of MP2, MP3, PCM and Vorbis.
Just to clarify it.
Other software that can passthrough or ignore video and convert audio to DTS and others is Hybrid.
XMedia Recode seems to be extremely buggy and completely useless to me. I tried to test it by converting a 5.1 channel DTS-HD MA (I believe this is lossless) audio stream to 5.1 channel DTS at 1536kbps, but the resulting audio was actually 2 channel stereo! Seemed kind of fishy when a 50 minute 6 channel lossless audio track took about 3 minutes to convert.
Then I tried to convert the same audio stream to AC3, but the highest bit rate on the drop down list was 640kbps, which is too low. It doesn't matter though, since the program will not let me select 640kbps or any other bit rate for AC3. No matter what I choose from the drop down list it simply defaults back to 32kbps! Why would I want to convert my lossless audio to 32kbps AC3??
Finally I tried to convert to AAC, which does have a bit rate option of 1536kbps, but there are some strange options that I don't understand. There are two options which are already ticked called TNS and Mid/Side. No tool tip or any explanation what they are. There is a drop down menu with no default selection, but has three options: Main, LC, LTP, again zero explanation what they are and no tool tip. When I click the help button it brings me to some German web page that I don't understand.
Again, thanks for the suggestion but I feel like I just wasted 20 minutes of my life I'll never get back.
Both DTS and AC3 are very old. DTS at 1536 is very inefficient, IMHO. At 16 bit FLAC will often achieve similar or lower bitrate.
- Compatibility means also acceptance by audiophiles and also hardware compatibility with inherent audio interfaces (S/PDIF) limitations.
- Audiophile codec Musepack is based on even older MPEG-1 Layer II so codec age never should be primary or secondary type of argument, older mean simpler techniques for lossy compression used, newer codecs techniques focus on delivering rather lower bitrate than higher quality audio when compared with previous generation.
True audiophiles don't recognize multichannel as real audiophile source of sound - this is direct outcome from statements of people claiming to be true and no compromise audiophiles.
Problem with AAC is lack of free (open source) sufficiently mature and high quality codec - available AAC codecs are usually limited.
From currently available (mean you can use them on your computer somehow) AAC encoders http://wiki.hydrogenaud.io/index.php?title=AAC_encoders , only two are considered as relatively audiophile - Apple and Franuhofer FhG .
Handbrake's AAC (avcodec) to encode 5.1 channel audio to 384kbps, which seemed like a good idea, but now that I think about it 64kbps per channel seems kind of low. Also, I don't know if this avcodec is a good one or not.
Maybe I will leave the audio I already encoded the way it is. It seems like too much hassle to go back and try to figure out which codec, which implementation of the codec, and what bitrate to use, and then have to re-encode all t he audio again. What would be an acceptable bit rate for the average person when encoding 5.1 channel audio using AC3 and/or AAC?
iam not an audiophile and am glad for it. When in past i converted audio separately from video, i used to use great program that is called LameXP. It supports Nero aac codec. And it has great results. I used it mainly for HEv2 SBR+PS conversion at 48kbps. Results are surprisingly great. The limitation of this codec is max 6(5.1) channels. But as iam not audiophile i can say, that default handbrake and vidcoder aac codec is very very poor. There on this site is how to make possible FDK in Handbrake. So thats only way to have good quality audio when encoding video and audio "at once". I'am now using avidemux and it has FDK codec as default, so no need to do audio and video separately.
Edit: usual bitrate for AC3 is 448 or 640kbps
Last edited by Bernix; 6th Jan 2018 at 12:52. Reason: Edit aac for ac3 :)
If you're open to learning some simple and pretty easy command line pipes, I would suggest demuxing with mkvtoolnix or mkvtoolnixgui, and then running the FLAC 5.1 file through ffmpeg to convert it to AC3 5.1 640 which I have been very happy with regarding quality. The advantage to using AC3 over FLAC or DTS would not only be a space savings, but that AC3 is compatible with pretty much everything. As to the AC3 encoder in Handbrake - someone correct me if I'm wrong - I thought it uses ffmpeg. The last time I had done any research on it, ffmpeg or Aften are supposed to be the recommended free AC3 encoders. If you're going to use AAC, demux the FLAC track from the file and run it through a program that uses qaac (the iTunes AAC encoder).
Last edited by stonesfan99; 7th Jan 2018 at 07:40.
5.1 AC-3 require for good quality at least 384kbps however 448kbps is recommended, if maximum compatibility (with DVD) is not your goal then you can encode AC-3 with even higher bitrate like 512kbs or 640kbps - last one is considered as perceptually lossless. I would no go for AAC 5.1 bellow 256kbps - as you can see there is no significant gain between AAC and AC-3 for most common listeners experience (case 320kbps AAC vs 384kbps AC-3). I see one situation where AAC have no alternative - content with more channels than 5.1 (e.g 7.1)- then AAC remain most sane combination - IMHO you can multiply number of audio channels by 48 (AAC) - 64 (AC3)kbps for each channel - this simple rule should be sufficient for very good quality encoding.
Thanks everyone for the suggestions. I ended up going with AC-3 at 640kbps. I used myFFmpeg to convert the original 5.1 channel DTS and FLAC audio to AC-3. The strange thing is that it went extremely fast. It converted a 20 minute 5.1 DTS track to 5.1 AC-3 in less than a minute. Is it normal for it to be that fast? myFFmpeg doesn't seem to have any quality settings for AC-3 other than the bit rate.
Just out of interest, can YOU hear any difference between your "original" 6 channel FLAC, the 6 channel AAC at 384 kbps that you originally converted them to in Handbrake, and the AC3 versions that you've finally settled on?
So often people - me included - can get so hung up on the numbers without actually viewing or listening to whatever it is, and saying "I'm happy with that." I honestly don't think that I can hear any difference between 5.1 AC3 at 640K and 448K. I might be able to hear a slight difference at 384K, but that could just as well be wishful thinking. So far, I've never noticed a difference comparing AAC and AC3 at the same bitrates. Sadly, due to my advancing age (I'm sure someone's added on at least a decade without me noticing! PAH!) my hearing isn't as good as it was when I was twenty, but it's still pretty good.
Not having a go, but just genuinely interested if the OP could actually hear the differences in his various encodings.
No, I can't hear a difference. Right now I'm using only my TV (stereo sound), so I couldn't hear a difference even if I tried. I want high quality sound because in the future I'll probably have a decent 5.1 channel audio system. Maybe I won't hear a difference on that either, who knows. Anyway, my question was whether it was normal for myFFmepg to encode audio so fast.
Ok. Like I said, I wasn't having a go.