IPB

Welcome Guest ( Log In | Register )

 
Reply to this topicStart new topic
recording level for wavs when final format is AC3?
krabapple
post Feb 3 2010, 23:32
Post #1





Group: Members
Posts: 2185
Joined: 18-December 03
Member No.: 10538



I'm transferring some multichannel analog recordings to multichannel .wav using Audition 1.0 and an M-Audio Delta 1010lt card. Using the card's 'consumer' input setting, and monitoring the recording with Audition's peak meter, the highest waveform peaks remain just below 0dBFS in any channel (~-0.5dBFS). The recording is 88khz/32bit then I output the file as a single 88/24bit interleaved multichannel .wav file. This becomes input into Audacity for conversion to AC3 using its FFmpg plugin.

Do I need to worry that these peaks could become 'overs' when the multichannel .wav is converted to AC3? IOW, should I be recording less 'hot' to account for possible overs from conversion? If so, what's a safe amount of headroom?

(FWIW, when I load my resulting AC3 file into Audacity , the peaks are still below 0dBFS. I presume this involves some sort of temporary conversion to .wav on the part of Audacity?)

(And , as I said I'm using Audacity's FFmpeg plugin to do the conversion. Try as I might, I cannot figure out how to get WAVtoAC3 software to work with the interleaved multichannel .wav file that Audition outputs. If anyone has gotten that situation to work, please let me know)

This post has been edited by krabapple: Feb 3 2010, 23:46
Go to the top of the page
+Quote Post
DVDdoug
post Feb 4 2010, 02:44
Post #2





Group: Members
Posts: 2542
Joined: 24-August 07
From: Silicon Valley
Member No.: 46454



QUOTE
Do I need to worry that these peaks could become 'overs' when the multichannel .wav is converted to AC3? IOW, should I be recording less 'hot' to account for possible overs from conversion? If so, what's a safe amount of headroom?
I don't claim to understand the "internals" of the AC3 encoder, but I assume it can handle "all 24 bits", and this shouldn't be a problem.

QUOTE
(FWIW, when I load my resulting AC3 file into Audacity , the peaks are still below 0dBFS. I presume this involves some sort of temporary conversion to .wav on the part of Audacity?)
All audio editors work with PCM data, which is a sequence of sample values similar to a WAV file. The main difference is that most audio editors use 32-bit floating point for internal/temporary storage, so there is a format conversion when you open or save.


QUOTE
Try as I might, I cannot figure out how to get WAVtoAC3 software to work with the interleaved multichannel .wav file that Audition outputs.
I haven't tried that... I've fed it separate files for each channel... And, I've only used 48kHz/16-bit. The funny thing is, the command line instructions for the Aften encoder (used by WAVtoAC3encoder) don't tell you how to use multiple files! BTW - Aften is "based on" FFmpeg, so I don't know how much difference there is.

How big is your WAV file? The WAV spec limits files to 2GB (or 4GB?). I don't have WAVtoAC3encoder on this machine, but I think there is a setting to ignore file size. (I'm usually working with "movie length" WAV files, and I use separate files to keep the file size in-spec.)

This post has been edited by DVDdoug: Feb 4 2010, 02:49
Go to the top of the page
+Quote Post
krabapple
post Feb 4 2010, 07:57
Post #3





Group: Members
Posts: 2185
Joined: 18-December 03
Member No.: 10538



QUOTE (DVDdoug @ Feb 3 2010, 20:44) *
QUOTE
Try as I might, I cannot figure out how to get WAVtoAC3 software to work with the interleaved multichannel .wav file that Audition outputs.
I haven't tried that... I've fed it separate files for each channel... And, I've only used 48kHz/16-bit. The funny thing is, the command line instructions for the Aften encoder (used by WAVtoAC3encoder) don't tell you how to use multiple files! BTW - Aften is "based on" FFmpeg, so I don't know how much difference there is.

How big is your WAV file? The WAV spec limits files to 2GB (or 4GB?). I don't have WAVtoAC3encoder on this machine, but I think there is a setting to ignore file size. (I'm usually working with "movie length" WAV files, and I use separate files to keep the file size in-spec.)



I keep my .wav filesizes below 2GB (to avoid all sorts of other problems), so filesize is not the issue. Unfortunately I've found the documentation for Aften/FFmpg/WAVtoAC3, such as it is, rather impenetrable. Luckily it's implemented in fairly no-brain fashion in Audacity, I just had to make sure the channel mapping was correct for my .wav files (it was).

(In the end AC3 conversion may be moot if I decide to use 'on the fly' AC3 encoding of the multichannel .wavs for SPDIF playback. That makes more sense than maintaining every mch file as a lossless and lossy version, as I'm doing now).





Go to the top of the page
+Quote Post
Alex B
post Feb 4 2010, 11:28
Post #4





Group: Members
Posts: 1303
Joined: 14-September 05
From: Helsinki, Finland
Member No.: 24472



I just tried creating a 5.1 wave file with Audition v. 3.0.1 and then encoding it with WAVtoAC3encoder v 4.4 (& aften v. 0.0.8). It created a 5.1 AC3 file just fine. I didn't check if the channel mapping was correct, but for instance foobar with the AC3 plugin has no problems in playing or converting the file.

This may be obvious, but have you actually placed aften.exe in the WAVtoAC3encoder's folder? Aften doesn't come with WAVtoAC3encoder. It is also possible that Adobe has fixed something after Audition v.1.0.

Regarding the peak level I'd use a bit lower volume, -1 - -2 dB or so. I don't think AC3 is any different than other lossy formats. For instance, most loud and compressed music tracks produce clipping on decoding to an integer bit depth when the format is MP3. Usually a -1.5 dB volume level reduction is not enough to prevent that completely (MP3 files can be adjusted in 1.5 dB steps). Such clipping seems to always have a very short duration, only a few samples at most. I don't think it can be audible if the other components in the signal chain can handle 0 dBFS without problems, but it might be a good idea to give the Dolby decoder, possibly enabled DSP effect processors and DA converter a bit easier time.

This post has been edited by Alex B: Feb 4 2010, 11:53


--------------------
http://listening-tests.freetzi.com
Go to the top of the page
+Quote Post

Reply to this topicStart new topic
1 User(s) are reading this topic (1 Guests and 0 Anonymous Users)
0 Members:

 



RSS Lo-Fi Version Time is now: 1st August 2014 - 13:18