It works fine here in winamp (vdub => file => save wav) . Note , ConvertAudioTo16Bit() is required for vdub
Code:vid= ffvideosource("audio.mxf") aud= ffaudiosource("audio.mxf") left = GetChannel(aud, 1) right = GetChannel(aud, 2) both=mergechannels(left, right) audiodub(vid,both) ConvertAudioTo16Bit()
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at the moment I don't have install it, however Vdub don't get me any error.
But if you give me your exactly ConvertAudioTo16Bit I can install as plugin.
I open your .wav but my winamp get me an error and don't open it.
While my output.wav is jerky -
Did you forget to disable the vdub audio filters from the last run ?
(make sure you use the .avs script only, don't do the downsample in vdub audio filters)
But if you give me your exactly ConvertAudioTo16Bit I can install as plugin.
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I uploaded the wrong file above; that was a 32 bit per sample (without ConvertAudioTo16Bit)
This is the correct one. If you compare the waveforms in audacity, your audio looks different (looks choppy on the Left audio channel)
If you are using vdub to output audio, audio=>direct stream copy will disable the filters -
mmm, there is not the flag on the "use advanced filtering" [see: AUset.bmp attached]
so I think Virtualdub don't use audio-filter. Or not? I'm not practically of vdub
so exactly what I have to do do disable the audio filters?
Analysing your code:
audiodub(vid,both)
ConvertAudioTo16Bit()
is there a way to do this functions with SounOut? -
SoundOut(output = "wav", filename="c:\outputFile.wav", atype=1
Sorry but sometimes I'm a [pc.bmp]
but with type=1 this is the results:
[winamp.bmp]
but maybe SoundOut is the solutionLast edited by marcorocchini; 20th Jul 2017 at 04:00.
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a fast look at the documentation would have shown,... typo: atype=1 -> type=1
congratulation you found it -
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vid= ffvideosource("v:\automazioneclip\input\audio.mxf" )
aud= ffaudiosource("v:\automazioneclip\input\audio.mxf" )
left = GetChannel(aud, 1)
right = GetChannel(aud, 2)
both=mergechannels(left, right)
audiodub(vid,both)
ConvertAudioTo16Bit()
SoundOut(output = "wav", filename="c:\outputFile.wav", type=1, format=0, autoclose=true, showprogress=true, overwritefile="yes")
=
[winamp.bmp]
... type=1 and format=0 what means? -
> ... type=1 and format=0 what means?
-> http://avisynth.org/mediawiki/SoundOut#WAV.2FAIF.2FCAF_Script_Parameters: -
vid= ffvideosource("v:\automazioneclip\input\audio.mxf" )
aud= ffaudiosource("v:\automazioneclip\input\audio.mxf" )
left = GetChannel(aud, 1)
right = GetChannel(aud, 2)
both=mergechannels(left, right)
audiodub(vid,both)
ConvertAudioTo16Bit()
But result remains the same [outputfile.wav]: is jerky -
If even the preview of the script is bad (play the .avs script in a media player or vdub), then it's more likely a decoding issue, not an ENcoding issue.
Which version of ffms2 are you using ?
Try changing FFMS2 builds, try vanilla, 32bit 2.17
http://code.google.com/p/ffmpegsource/downloads/detail?name=ffms-2.17.7z -
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btw.: Why not simply use ffmpeg for the whole thing?
Code:ffmpeg -i h:\AUDIO.MXF -vn -filter "channelmap=1|2:channel_layout=stereo" h:\out.wav
seems to work fine here:
Code:Input #0, mxf, from 'h:\AUDIO.MXF': Metadata: uid : 38d41b00-3164-05c1-0270-08004602023b generation_uid : 38d41b00-3164-05c2-0270-08004602023b company_name : SONY product_name : Opt product_version : 1.41 product_uid : 060e2b34-0401-0103-0e06-012002010300 modification_date: 2013-05-19 20:15:46 timecode : 00:46:20:01 Duration: 00:00:06.68, start: 0.000000, bitrate: 62578 kb/s Stream #0:0: Video: mpeg2video (4:2:2), yuv422p, 720x608 [SAR 152:135 DAR 43], 25 fps, 25 tbr, 25 tbn, 50 tbc Stream #0:1: Audio: pcm_s24le, 48000 Hz, 4.0, s32, 4608 kb/s Output #0, wav, to 'h:\out.wav': Metadata: uid : 38d41b00-3164-05c1-0270-08004602023b generation_uid : 38d41b00-3164-05c2-0270-08004602023b company_name : SONY product_name : Opt product_version : 1.41 product_uid : 060e2b34-0401-0103-0e06-012002010300 modification_date: 2013-05-19 20:15:46 ISMP : 00:46:20:01 ISFT : Lavf55.7.100 Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, stereo, s1, 1536 kb/s Stream mapping: Stream #0:1 -> #0:0 (pcm_s24le -> pcm_s16le) Press [q] to stop, [?] for help size= 1253kB time=00:00:06.68 bitrate=1536.1kbits/s video:0kB audio:1252kB subtitle:0 global headers:0kB muxing overhead 0.007797%
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Poison is there a way to generate directly the output.wav file by launching the script?
e.g.
VirtualDub.SaveWAV("o:temp.wav"); -
you are getting a virtual dub error when using ffmpeg -> you are doing something wrong,..
attached the out.wav that was created,... by the ffmpeg call,.. -
nice selur - I think that is fairly new in ffmpeg . Audio handling used to be pitiful in ffmpeg
But there is still some audio difference? Compare with ffms2 and avisynth stereo mix
It shouldn't be, because ffms2 is based on ffmpeg....
I'm not expert on audio, but the avisynth mixed sounds "fuller" ?? There are waveform difference in audacity as well
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