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31st August 2006, 11:41 | #201 | Link | |
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I think that it sould be possible to obtain a good approximation of dialog level using exponents values (weighted according to freq band) |
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31st August 2006, 18:00 | #202 | Link | |
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- This force to 2 pass encoding, and not live encoding. - Automatic Dialog Normalization can work maybe with modern music, but never with movie tracks. The correct DialNorm must be calculated over a selected fragment with dialogs presents, not over the full track with aleatory silences. - There are enough tools to calculate this parameter over the source to be encoded, and the related DRC type (film, music, speech, ...) I don't know how can be selected automatically. |
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31st August 2006, 20:40 | #203 | Link | |
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Like Tebasuna said, the dialog level should be measured using just an excerpt which typifies the dialog level of the whole stream. I can modify the wavrms utility program to accept a time or sample range and improve the dialnorm calculation as well...and/or include it in Aften as a 2-pass option. A possible solution for live encoding might be to add some sort of "dialog calibration" functions in libaften. Something like start_calibration(), append_calibration(audio samples), end_calibration(). This could be used by a live encoder app to do a sort of microphone test. Or it could be used by a production app by passing a user-selected dialog range. Extending that idea to modifying existing AC3 streams, it could optionally take ac3 frames as input and analyze the exponents. I'm just throwing ideas out here... I need to get the DRC actually working before jumping into any of these. |
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1st September 2006, 02:48 | #204 | Link |
interlace this!
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with live stuff, the level should already be sorted before it hits the encoder
so i can't see it hurting to just enter a number in the CLI, then let the DRC handle the rest. if it's a little off it shouldn't hurt too bad. considering other encoders rely on user input, there's no reason aften can't too perhaps a "-live" switch or similar could be added so aften knows how many passes to do? or perhaps 2-pass should be specified explicitly (that's probablythe way to do it).
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1st September 2006, 15:39 | #205 | Link | |
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At the Paris AES convention, some people from Swedish TV explained (roughly) how they are proceding: DRC is computed on the fly, while dialog level is selected based on the content type (a set of standard values for movies/news/sports/advertising/...). If we assume that this represent a typical use case, then what is needed is dynamic DRC and ability to manually specify dialog level (even while encoding). |
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2nd September 2006, 00:51 | #206 | Link | |
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13th September 2006, 03:45 | #209 | Link |
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I´ve made a small batch file that uses the aplication wavrms.exe presents on Aften v0.05 package.
With this batch we can use Aften with a "automatic measure of parameter Dialog Normalization". Code:
rem This batch works on Win-XP @echo off setlocal enabledelayedexpansion wavrms.exe video1.wav | find "Dialnorm" > "Dialnorm.txt" for %%F in (Dialnorm.txt) do ( for /F "usebackq tokens=1 delims=B" %%J in ("%%F") do ( set char=%%J set char=!char:~12,2! echo Dialnorm = - !char! dB aften -b 224 -dnorm !char! -acmod 2 -bwfilter 1 -dcfilter 1 video1.wav video1.ac3 ) ) |
13th September 2006, 07:44 | #210 | Link | |
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hi ADLANCAS
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so much. ps: are you from SP,MG,RJ,CE,RS...? (BR here too) regards. |
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14th September 2006, 23:46 | #212 | Link |
Huh?
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What is currently being worked on?
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15th September 2006, 01:58 | #213 | Link |
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DRC, dialnorm, and speed. Also, I'm trying to make up my mind on changing the build system.
I just applied a simplification of the dialnorm calculation in wavrms. Instead of using the sort/percentile approach as in replaygain, I chose something a bit simpler. Now it does averaging of RMS, but throws out values which are outside of a reasonable dialog range. Some testing has shown that it works pretty well. I'd really like to get some feedback on it though. Since it was applied today, it will show up in tomorrow's daily build. -Justin Last edited by jruggle; 15th September 2006 at 02:01. |
15th September 2006, 02:05 | #214 | Link |
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You could see his blog:
http://aftenblog.blogspot.com/ Last edited by ADLANCAS; 15th September 2006 at 02:13. Reason: Oh, sorry! I posted in the same time as jruggle posted. |
15th September 2006, 02:14 | #215 | Link |
Huh?
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Oh, I forgot about the blog . Justin, do you think channel coupling might be a possibility in the future?
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16th September 2006, 21:06 | #216 | Link | ||
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@ jruggle,
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thanks. |
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19th September 2006, 18:18 | #217 | Link | |
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I know you probably have a long TODO list but... Any chance that you have made any progress on inline resampling? Like from 44100 to 48000. |
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19th September 2006, 21:04 | #218 | Link |
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Aften is looking great so far, but I have a file on which it generates bad output. The wav source is this 20 second excerpt:
http://members.home.nl/meuwissenth/0.0.0.wav Which sounds just fine. Once I run it through Aften with the parameters -b 192 -acmod 2 or even just either one of them, it outputs this: http://members.home.nl/meuwissenth/0.0.0.ac3 Which plays back with pauses and the seeker bar visibly skipping in VLC Media Player. Am I doing something wrong here? |
20th September 2006, 02:15 | #219 | Link | |
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Azid and the new NicAudio (is a 44100 Hz) can decode the 0.0.0.ac3 without problems. The same wav resampled to 48000 play ok with VLC, and other ac3 44100 Hz have the same problem in VLC. Edit: my vlc version is 0.8.2 |
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20th September 2006, 03:54 | #220 | Link |
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Yes, sort of. The basic RMS calculation is still there. Replaygain does not have anything to do with dialog level though, it has to do with _overall_ perceived loudness. The 95% point does a good job for that measurement, but for dialog in a movie, for example, it isn't as good. I was adjusting it down by a fixed amount as an educated guess based on sample data...this really was not very accurate though. The new method throws out very quiet and very loud sections, which can occur pretty often in movies, but rarely include normal dialog. Also, the specification of a time range gives the user more control to select a good section of audio which has typical dialog.
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