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| dloneranger |
| Posted: Aug 26 2012, 04:48 PM |
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Moderator
Group: Moderators
Posts: 2366
Member No.: 22158
Joined: 26-September 07
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I'm rewriting WavNormailze, to cope with wave64's but I'm having trouble with the chunk headers of the wav file virtualdub outputs
They just seem a bit off
Here's the hex dump of the start of the file
The file length is 4226109992, $FBE55228
| CODE | Offset 0 1 2 3 4 5 6 7 8 9 A B C D E F
00000000 72 69 66 66 2E 91 CF 11 A5 D6 28 DB 04 C1 00 00 riff.‘Ï ¥Ö(Û Á
00000010 28 52 E5 FB 00 00 00 00 77 61 76 65 F3 AC D3 11 (Råû waveó¬Ó
00000020 8C D1 00 C0 4F 8E DB 8A 66 6D 74 20 F3 AC D3 11 ŒÑ ÀOŽÛŠfmt ó¬Ó
00000030 8C D1 00 C0 4F 8E DB 8A 28 00 00 00 00 00 00 00 ŒÑ ÀOŽÛŠ(
00000040 01 00 02 00 80 BB 00 00 00 EE 02 00 04 00 10 00 €» î
00000050 64 61 74 61 F3 AC D3 11 8C D1 00 C0 4F 8E DB 8A dataó¬Ó ŒÑ ÀOŽÛŠ
00000060 C0 51 E5 FB 00 00 00 00 00 00 00 00 00 00 00 00 ÀQåû
00000070 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00
00000080 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00
00000090 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 |
So, there's the riff header at 0, with length $FBE55228
(start+length) rounded up to next multiple of 8 = end of file+1, as expected
fmt chunk at $28, with length $28
(start+length) rounded up to next multiple of 8 = $50, as expected, the start of the data chunk
data chunk at $50, with length $FBE551C0
(start+length) rounded up to next multiple of 8 = $FBE55210, ummm now that's not what I expected
That is the (end of file+1) minus $18
There's no addition chunks at that position and as far as I can make out reading the specs, it should be equal to end of file+1
Am I interpreting the data incorrectly? Or is the data length wrong?
It's exactly off by the length of the chunk header (guid+int64)=$18
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MultiAdjust JoinWav WavNormalize FFMPeg Input Plugin v1827 UnSharpMask
Windows7/8 Codec Chooser
All FccHandlers Stuff inc. Installers for acm codecs AAC, AC3, LameMp3 |
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| dloneranger |
| Posted: Aug 26 2012, 07:58 PM |
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Moderator
Group: Moderators
Posts: 2366
Member No.: 22158
Joined: 26-September 07
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If I've got this right then in AVIOutputWAV.cpp
void AVIOutputWAV::WriteHeader(bool initial)
in the block starting with
if (mbWriteWAVE64) {
| CODE |
if (!initial) {
*(uint64 *)&dwHeader[4] = mBytesWritten;
}
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should be
| CODE |
if (!initial) {
*(uint64 *)&dwHeader[4] = mBytesWritten+24;
}
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It didn't really help though, as part2 was wave64->aac, and I can't find an aac encoder that accepts wave64's
..... sigh......
Nevermind, I'll have to add a 'Save wav in just under 2Gb chunks' function to vdub
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MultiAdjust JoinWav WavNormalize FFMPeg Input Plugin v1827 UnSharpMask
Windows7/8 Codec Chooser
All FccHandlers Stuff inc. Installers for acm codecs AAC, AC3, LameMp3 |
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| phaeron |
| Posted: Sep 2 2012, 09:17 PM |
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Virtualdub Developer
Group: Administrator
Posts: 7773
Member No.: 61
Joined: 30-July 02
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Yes, this is a bug. Unlike regular RIFF, WAVE64 lengths include the chunk header. (Don't know why people do this in file formats... seems useless.) I'll fix it for 1.10.3.
As for your issue with AAC encoders that accept large streams, some do accept oversize WAV files if they are in stdin mode. This is used for live piping of data from one program to another and causes the encoder to ignore the main data length in the header.
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| dloneranger |
| Posted: Sep 2 2012, 09:32 PM |
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Moderator
Group: Moderators
Posts: 2366
Member No.: 22158
Joined: 26-September 07
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stdin's a pain - there's all the passing # of channels, bitrate etc which doesn't lend itself to simple batch files
what I ended up doing was to modify WavNormalize to accept wav64 files and spit out multiple 1.9Gb files
then in the batch file, scan for multiple parts and pass them to neroaac as it's input
shame really, wav64 is such a simple modification I'm surprised it's not more widely accepted
the changes to the header lengths and padding alignment are just there to make life interesting for us.....
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MultiAdjust JoinWav WavNormalize FFMPeg Input Plugin v1827 UnSharpMask
Windows7/8 Codec Chooser
All FccHandlers Stuff inc. Installers for acm codecs AAC, AC3, LameMp3 |
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