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30th August 2011, 00:02 | #401 | Link |
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After the HydrogenAudio listening test determined that fhgaac was superior to Nero AAC I decided to do some tests:
1. Speed is pretty much the same. FHGAAC is a tad slower than Nero (17 min vs. 16 min), but this is hardly significant. 2. Since FHG does not support ABR (and CBR is not really desirable), quality mode looks like the optimal mode to do a comparison. But this is not so easy: Nero has a quality scale from 0.00 to 1.00. FHG uses VBR presets in the range of 1 to 5. LameXP only has a slider for the Nero style quality settings, so it somehow maps the Nero settings to the FHG settings. In my first test I used a quality setting of 0.30 (which roughly corresponds to an average bitrate of 96 kbp/s). For Nero I got a file size of 67.2 MB, for FHG the file was much smaller (49.1 MB). The 0.30 quality setting was mapped to FHG VBR 2. The next encode with FHG was done with a quality setting of 0.40 which corresponded to a FHG setting of VBR 3. This time the file size was almost identical to the encode done with Nero using a quality setting of 0.30. I do not have the means to test the quality of the encoded files, my audio hardware is just too shabby. All encoded files sounded pretty much identical to me. But it looks like the mapping of the Nero quality settings to the FHG settings is not perfect yet. Another hint: If you want to use the FHG encoder but you do not care for WinAmp, you can extract the necessary files from the installer executable without actually installing WinAmp. Download the WinAmp installer, open it with 7zip and extract the required files. Cheers manolito |
30th August 2011, 00:55 | #402 | Link | |
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As the quality parameter of the FHG encoder only accepts integers, the values have to be rounded. That makes the mapping kind of coarse-grained... I know. But I don't think it would be very "fair" to tell people to extract the encoder plugin without actually installing Winamp
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30th August 2011, 23:23 | #403 | Link | |
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Bug normalizing multichannel wav files
While testing the fhgaac encoder I stumbled upon a severe bug which probably has been there for quite a while:
After converting a 6-ch AC3 to a 6-ch AAC I noticed that the volume of the encoded file was way lower than the volume of the original AC3. Then I did the same encode using Nero, but the volume was still way too low. So I figured that this was a AAC issue, and I thought I might get better results using LameXP's normalize feature. But it turned out that this approach did not work at all. SoX seemed to do its thing normally, but afterwards the encode stopped immediately. No matter if Nero or FHG was chosen as the encoder. Looks like the SoX output is not usable for ther AAC encoders. Here is the log: Quote:
Cheers manolito |
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30th August 2011, 23:37 | #404 | Link | |
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Quote:
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30th August 2011, 23:48 | #405 | Link | |
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http://lamexp.git.sourceforge.net/gi...=HEAD#71a113b0 @manolito The reason why the audio converted from the AC-3 source appears to have a lower volume is because AC-3 has much bigger dynamic range. Also I can confirm that there is a problem with the normalization filter. It's not only the AAC encoder. The same issue occurs with OggEnc, for example. I will have to investigate what is going wrong here. But I will be away for the next few days, so please be patient...
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4th September 2011, 22:52 | #407 | Link |
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Am I supposed to know what a "concentric" folder is?
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4th September 2011, 23:40 | #408 | Link |
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RE normalization filter:
This has nothing to do with the decoder or encoder which LameXP uses. It is solely a SoX issue. And the problem was already present in V. 4.02 final. For a test just try to convert a 6ch WAV (does not matter if it is in the Extensible Format or not) to a normalized 6ch WAV. Fails... Cheers manolito |
4th September 2011, 23:42 | #409 | Link | |
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Can this be reproduced with the "official" SoX binaries for Win32?
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5th September 2011, 02:12 | #410 | Link | |
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in a simple way "one inside another" but I think "recursive" can best be understood by a dev.
ie a folder "Micky Maus" a folder "Goofy" into "Micky Maus" an mp3 into "Goofy" re normalization filter there is an WaveGain(S) too Quote:
Last edited by Motenai Yoda; 5th September 2011 at 02:45. |
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5th September 2011, 02:55 | #411 | Link | |
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(Seems to work as expected for me) The "prepend relative source file path to output file" option does not apply, if "save output files to the same location where the input file is located" is enabled.
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5th September 2011, 13:46 | #412 | Link |
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1- ok but I would expect it was the default method for drag'n'drop
2- look http://www.youtube.com/watch?v=MwyXFIObdug Last edited by Motenai Yoda; 5th September 2011 at 13:48. |
5th September 2011, 14:09 | #413 | Link | |
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It's implemented that way, because adding a folder recursively might add a HUGE number files which can take a very long time. Maybe the default behavior can be changed in a way that will trigger "recursive" mode when a folder that only contains sub-folders (but no files) is dropped. (I want to avoid adding yet another option for this)
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5th September 2011, 15:26 | #415 | Link |
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Thx for the info. At least it's not a build issue then
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5th September 2011, 18:13 | #416 | Link |
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also broken with 14.3.1 or 14.3.0...
_
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5th September 2011, 22:32 | #417 | Link |
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This evening I did some more tests: SoX' gain filter doesn't fail with all multi-channel sources. And it only fails on some when the "-n" option is used
Anyway, if I also add the "-e" or "-b" option that seems to fix the issue. At least for my samples that failed before. Consequently LameXP will now use "-ne" instead of "-n" only. Hopefully it works with your sample too. A new build is available via auto-update...
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5th September 2011, 23:39 | #418 | Link |
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Yes, this fixes it...
It adds a little to the encoding time, but I could not break it using all kinds of multichannel input files. I still find it hard to explain why omitting the -e or -b option causes SoX to completely screw up the conversion. Strange... Anyway, it works now, thanks again. Cheers manolito |
6th September 2011, 20:30 | #419 | Link | |
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-ne will rise the loudness of the central channel if its peak is lower than the side channels or will rise the loudness of the side channels if its peaks is lower than the central channel... in other words the some level of DRC will be applied... _
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6th September 2011, 21:04 | #420 | Link | |
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As I am not a SoX developer, this is not in my hands, unfortunately.
(And nope, I'm not currently planning to add yet another tool to LameXP for normalization) Quote:
Anyway, generally I would assume that all channels had been recorded at the same levels. And if not, that's probably unintentionally. So their peaks should be at the same volume too! A channel would only be "amplified" more than the others, if there is not a single peak in that channel for the whole duration of the audio... Might be applied.
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Tags |
aac, aotuv, flac, lame, lamexp, mp3, mp4, ogg, oggenc, opus, vorbis |
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