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17th August 2011, 07:13 | #11182 | Link | |
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regsvr32.exe ASAudioHD.ax Last edited by nibus; 17th August 2011 at 12:00. |
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17th August 2011, 11:02 | #11184 | Link | ||
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edit: Here it is- http://forum.doom9.org/showthread.ph...84#post1469584 Quote:
Last edited by nibus; 17th August 2011 at 11:23. |
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17th August 2011, 13:52 | #11185 | Link |
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I know we're not supposed to ask where to get these filters, but I'm going to just ask rhetorically...where are we supposed to find all these random older versions of this filter? Is there one version that works perfectly with every possible layout? Someone really needs to get on the ball and get DTS-HD decoding into libavcodec.
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17th August 2011, 14:07 | #11186 | Link |
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1.1.0.0 work also with non-standard channel mapping without garbage.
Like don't exist channels Ls-Rs in WAV make a remux to BL-BR correct to my opinion.
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17th August 2011, 15:33 | #11187 | Link |
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It's almost getting to be more worth the time to just fake-header a DTS-HD file and let the DTS suite decode the files. I really find that having to switch filter versions to be inexcusable (on Arcsoft's part, not madshi, he makes awesome stuff). I have no idea how I would make a 6.1 wav, though. There is no documentation on channel masks for wavavimux.
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17th August 2011, 18:14 | #11188 | Link | ||
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CHANNEL NAME --- Decimal Value ŻŻŻŻŻŻŻŻŻŻŻŻ ŻŻŻŻŻŻŻŻŻŻŻŻŻ FRONT_LEFT 1 FRONT_RIGHT 2 FRONT_CENTER 4 LOW_FREQUENCY 8 BACK_LEFT 16 BACK_RIGHT 32 FRONT_LEFT_OF_CENTER 64 FRONT_RIGHT_OF_CENTER 128 BACK_CENTER 256 SIDE_LEFT 512 SIDE_RIGHT 1024 ************************************* TOP_CENTER 2048 TOP_FRONT_LEFT 4096 TOP_FRONT_CENTER 8192 TOP_FRONT_RIGHT 16384 TOP_BACK_LEFT 32768 TOP_BACK_CENTER 65536 TOP_BACK_RIGHT 131072 RESERVED 262144 |
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17th August 2011, 18:48 | #11189 | Link |
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I make a mono wav that contains silence, make it substantially longer than the track we're trying to work with (we'll call this "the useless source") I need to decode, encode a track with the exact specs as the useless source, copy the first 140 bytes in a hex editor, not the last character and what its pattern of character looks like before that character, in the useless source, cut up to the last character of our nice header, and copy INTO the beginning of the useless source. It's very time consuming, but since it's a DTS tool, and all things most likely come from the tool, there can't be anything it can't handle.
Now I see the scheme for the channel mask, it's just sums! I never saw the chart you posted though, so thank you! Last edited by ramicio; 17th August 2011 at 19:13. |
19th August 2011, 05:00 | #11190 | Link |
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Would there ever be any chance of taking multiple mono WAV files in to create a single multi-channel file? I can't find any decent and sure-fire way to do this with any other utility. WavAVImux to graphedit with wavdest, you are limited to a 4 GB output. Avisynth and VirtualDub, save WAV, can break the 4 GB barrier, but what about weird channel mappings such as 6.1? The there's wavewizard which I can't even begin to figure out, and have no idea about size limitations or channel mapping.
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19th August 2011, 05:56 | #11191 | Link | ||
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^ wavi.exe:
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WAVI v1.06F - (c) 2k7 Tamas Kurucsai, with lots of help from tebasuna51 Licensed under the terms of the GNU General Public License. This utility extracts the first uncompressed PCM audio track from an AVI file and saves it to a WAV file. This is not quite useful for most AVI files since they usually contain some kind of compressed audio, but it can come handy when it's needed to save an audio track from AviSynth. Usage: WAVI <avi-file> [ <wav-file> [ /R | /X | /M <mask> ] ] If <avi-file> is a valid AVI file which contains PCM audio, the audio track will be written to <wav-file> as a WAV file. If '-' is passed as <wav-file>, the WAV file will be written to the standard output. If <wav-file> is not given, only information will be printed about the audio track. On success, the exit code will be 0 and the first line printed to the standard error will look like the following: Found PCM audio: <c> channels, <r> Hz, <b> bits, <l> seconds. where <c> is the number of audio channels, <r> is the sampling rate, <b> is the number of bits per sample and <l> is the length of the track in seconds. If the audio track contains floating-point samples, the next line printed to the standard error will be: Audio track contains floating-point samples. If an error occurs, the exit code will be 1 and some useful error message will be printed to the standard error. WAV files larger than 4 GB may be created. However, such WAV files are non-standard and may not be handled correctly by some players and encoders. A warning will be printed to the standard error when such a WAV file is created. WAVI accepts the following options: /R - Write a raw file of samples without the WAV header. /X - Write an extended WAV header containing the default channel mask for multi-channel audio. /M <mask> - Write an extended WAV header containing the specified channel mask for multi-channel audio. The default channel masks are: Mask Chan. MS channels Description ---- ----- ------------------------- ---------------- 4 1 FC Mono 3 2 FL FR Stereo 259 3 FL FR BC First Surround 51 4 FL FR BL BR Quadro 55 5 FL FR FC BL BR like Dpl II (without LFE) 63 6 FL FR FC LF BL BR Standard Surround 319 7 FL FR FC LF BL BR BC With back center 255 8 FL FR FC LF BL BR FLC FRC With front center left/right Some other common channel masks: Mask Chan. MS channels Description ---- ----- ------------------------- ---------------- 7 3 FL FR FC 263 4 FL FR FC BC like Dpl I 271 5 FL FR FC BC LF 59 5 FL FR BL BR LF Quote:
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19th August 2011, 09:53 | #11192 | Link |
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WaveWizard works fine for me but don't fill the MaskChannel field (always 0) then you need some other util like:
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wavfix Usage: WavFix <input.wav> [output.wav] [-ignorelength] [-o #] [-m #] [-i/f #] [-c #] [-s #] Where: <input.wav> wav or w64 up to 8 channels, int or float, 8-16-24-32-64 bits, PCM/WAVE_FORMAT_EXTENSIBLE/W64, any extra chunks, allowed > 4GB. To use STDIN use - as input filename. [output.wav] If not present, input.wav is used suffixed with: _fix The header can be selected, see -o parameter. [-ignorelength] If present the length in wav header is ignored, useful for wav > 2/4 GB. Problem: extrachunks at end of file treated as data [-o #] Output header. 0=RAW, 1=WAV(Default), 2=RF64, 3=W64 [-m #] MaskChannel. Force output WAVE_FORMAT_EXTENSIBLE. 0 to default If any of the next parameters are present the input is considered RAW: [-i #] BitsPerSample Integer. Default 16. Valid 8, 16, 24, 32 [-f #] BitsPerSample Float. Valid 32 or 64. [-c #] NumChannels. Default 2. Valid only 1 to 8. [-s #] SampleRate. Default 48000 Hz. Any value is allowed.
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19th August 2011, 14:55 | #11193 | Link |
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Wavi doesn't work for me once an AVI file is created with wavavimux, even for ones that are under 2 GB. It used to, now it doesn't. It simply says there is audio in there, but it's not PCM. I don't understand how to use wavewizard, and I don't understand what that wavfix thing would even do. I just don't get how more people, or anyone at all has asked about eac3to being able to accept multiple mono wavs as input.
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19th August 2011, 18:23 | #11194 | Link | |
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<intermed.avs> DirectShowSource("audio-only.avi") </intermed.avs> wavi intermed.avs final.wav -M ### Last edited by Midzuki; 19th August 2011 at 19:56. Reason: grammar; overdue correction;; argh;;; |
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19th August 2011, 18:27 | #11195 | Link |
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Because, as previously stated, I have problems with certain channel schemes of DTS-HD Master, and I'm not going to try to hunt down the internet for specific versions of filters that I'll never find. So I must use the DTS suite to decode these weird ones into mono wavs. My end goal is a 5.1 FLAC. So I need to feed eac3to a multichannel WAV to downmix into 5.1 channels.
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19th August 2011, 19:18 | #11196 | Link |
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Ah, I see, at last
Hmmm, IMHO there really is no good reason why eac3to doesn't support .AVI as input. It's a shame that neither sox, nor wavi, can output .W64 or .RF64 Last edited by Midzuki; 20th August 2011 at 00:23. |
19th August 2011, 21:18 | #11198 | Link | |
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How want you downmix the Ls,Rs,Lsr,Rsr to only 2 Surround channels? Please put the coefficients X1, X2 in: SL = X1 x Ls + X2 x Lss SR = X1 x Rs + X2 x Rss I can make an avs file with your mix and can be used by BeHappy (or wavi) to obtain the 5.1 flac
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19th August 2011, 21:26 | #11199 | Link |
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I'm fine with the way eac3to handles 7.1 stuff. Never had a problem. It's these pesky 6.1 I come across now and then. I like eac3to because it detects clipping. Would it be possible for avisynth to turn out 6.1?
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20th August 2011, 03:19 | #11200 | Link |
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Does ffdshow decode 6.1 DTS-HD? If so you could open the DTS file via DirectShow, use ffdshow's mixer matrix to mix it to stereo, 5.1ch, or any way you like. From there, any encoder which will encode via DirectShow should be able to do the job.
I sometimes use ffdshow to mix to stereo (for those times when the audio track starts off stereo then switches to 5.1ch, and foobar's internal 5.1ch mixer has a hissy fit), while using foobar2000 to encode to MP3. Assuming ffdshow decodes 6.1ch DTS, I can't think of a reason why you couldn't use ffdshow to mix it to 5.1, then use foobar2000 to encode the output directly to flac. Or, if you don't want to use ffdshow's mixer, you could leave it disabled and get foobar200 to spit out a multichannel wave file. You'd need the foobar DirectShow decoder plugin. |
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