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Old 18th March 2012, 00:00   #1  |  Link
wonkey_monkey
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Cutting/editing audio - Avisynth mangling it?

Hi all,

I've just come across this oddity. I've cut a clip from a DVD rip in two using trim, then spliced the two clips back together with +. Using audiograph I can see the waveform of the audio, and on first stepping through the frames in VirtualDub (so not playing audio) it looks like this:



But after playing back any of the audio (before, during, or after the splice), it becomes this:



And the glitch is audible. Reloading the AVS file resets it again.

Can anyone shed some light as to what's going on, and tell me if there's any way to avoid this?

David
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Old 18th March 2012, 01:14   #2  |  Link
hartford
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Most interesting. I've not seen this before.

What version of Avisynth are you using?
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Old 18th March 2012, 01:34   #3  |  Link
poisondeathray
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Try alignedsplice ++ instead of unalignedsplice +
http://avisynth.org/mediawiki/AlignedSplice

If you have VBR audio use EnsureVBRMP3Sync
http://avisynth.org/mediawiki/EnsureVBRMP3Sync
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Old 18th March 2012, 09:39   #4  |  Link
wonkey_monkey
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My AviSynth version is 2.58. I tried ++ and the same thing occured. The audio is AC3, loaded with nicac3source.

David
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Old 18th March 2012, 09:49   #5  |  Link
wonkey_monkey
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Strangely (at least it seems strange to me) EnsureVBRMP3Sync may have "fixed" it. On the downside, it only works if I have Audiograph in the chain, otherwise it refuses to play (VirtualDub just sticks at "Dub in progress" but doesn't progress). It also seems to make Audiograph even more unstable.

Davd
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Old 18th March 2012, 16:41   #6  |  Link
wonkey_monkey
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More info: it seems I don't have to play any audio at all. If I step through to the cut in VirtualDub, the first time I go by it shows the clean join, the second time (either forwards or backwards, and as long as I go far enough [to get it cleared from a cache?]) it's glitched.

I wonder if it's a problem with nicac3source - but then how would nicac3source know to return anything other than the correct unglitched audio when it's requested? Would it "know" that the audio is at the start/end of a clip and fade it like this?

David
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Old 18th March 2012, 16:46   #7  |  Link
poisondeathray
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Quote:
it only works if I have Audiograph in the chain, otherwise it refuses to play (VirtualDub just sticks at "Dub in progress" but doesn't progress)
was it 6ch audio? I think vdub has problems playing 6ch audio through avs script

but the same video loaded directly works, or playing .avs in media player works, or if you downmix to stereo it works

it doesn't seem to matter how it's loaded (different source filters, nicac3source, ffaudiosource, etc.... or if you use separately with audiodub() )

this seems to occur on a variety of different video samples, and at least a few people have experienced similar results, so I think it's a vdub bug

see this discussion
http://forum.videohelp.com/threads/3...dio-in-preview

Last edited by poisondeathray; 18th March 2012 at 16:54.
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Old 18th March 2012, 17:35   #8  |  Link
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It is initially 6ch AC3, but the first thing I do after loading it is a getchannel(3,3) to have just the center channel (while still keeping it stereo for later mixing). In this case it just seems to have been EnsureVBRMP3Sync (without audiograph) that broke it.

David
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Old 18th March 2012, 21:08   #9  |  Link
IanB
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This looks like the Audio cache stuff up fixed at version 1.33 of cache.cpp, "Bugfix AudioCache snafu (Gavino)".

Try the AviSynth 2.6.0 Alpha3 [May 25th] which includes this fix.
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Old 18th March 2012, 22:19   #10  |  Link
wonkey_monkey
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That seems to have sorted it out, thanks Ian!

David
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Old 14th August 2012, 19:34   #11  |  Link
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Just to report I have just spotted the same glitch (using the same clips as my last report, regardless of + or ++) happening even with 2.60 Alpha 3 - on first view there is no visible splice, but after advancing (but not rewinding) more than about 7 frames, then returning, the splice is visible/audible.

(I'm currently trying to reproduce the problem using blankclip and tone so it's easily reproducible, but haven't had any luck yet)

If I was running my script through Avidemux via AVSproxy, for example, would this bug be avoided because none of the audio would have been cached? I'm really hoping this to be true because my project is now nearly 15 minutes long with lots of tricky cutting!


Edit: it seems this is occurring throughout my project wherever I've cut/spliced a clip - which is a lot. Is there any way to disable audio caching entirely - if that's practical - even if I have to compile my own DLL?

David

Last edited by wonkey_monkey; 14th August 2012 at 20:48.
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Old 15th August 2012, 00:18   #12  |  Link
IanB
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The cache if it is working correctly should make no difference, apart from performance. If the audio sample number 123456 has a value of 42 then each and every time you fetch that sample the value must be 42.

The problem fixed at version 1.33 of cache.cpp, meant that under some conditions the buggy version of the cache mistakenly returned samples from a different part of the clip, catastrophic, yes.

The other issue with the cache is how it complicates requests to an audio source filter. In a simple linear only access script it is very simple, every audio sample is requested once only and in order. With Trim's, MergeChannel's and other audio processing the situation can become very complex. The source filter will see requests for audio samples in chunks and in an order determined by the topology of the script audio processing graph.

The cache can suppress some of these requests if that data has previously been requested. When the cache buffer is not big enough to satisfy all requests for duplicate data then the source filter will see partial duplicate requests for only the uncached data samples i.e. cache misses. The audio cache is self tuning, it initially starts in the OFF state. When it receives requests for data that have been previously requested it enters the ON state with a modest buffer size. In the ON state it monitors for partial cache miss events, when they occur the cache increases it's size. So initially the source filter will see full duplicate requests, then some partial duplicate requests and eventually only initial requests that reflect overall gross scene trimming and order. If the source filter cannot reliably seek and return the identical samples as it did for the first time request then things can indeed be very dire.

The EnsureVBRMP3Sync() filter does 2 things, one it adds an initial 1MB audio cache, and two it enforce a linear order of access for requests to the source filter. Any backwards order requests result in a seek to the very beginning followed by a series of slow linear requests up to the required access point.

If you want to roll your own avisynth.dll without audio cacheing. Edit src\core\cache.cpp look for Cache::GetAudio. After the range checking (before zero, after end of clip) and protection code add a return child->GetAudio(start, count, env); Good luck.
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Old 15th August 2012, 17:06   #13  |  Link
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Quote:
The other issue with the cache is how it complicates requests to an audio source filter. In a simple linear only access script it is very simple, every audio sample is requested once only and in order. With Trim's, MergeChannel's and other audio processing the situation can become very complex. The source filter will see requests for audio samples in chunks and in an order determined by the topology of the script audio processing graph.
Ah, so it could be that the non-linear access is confusing nicac3source, rather then it being a problem with AviSynth - which might also explain why I haven't yet found the issue on sources where I've used ffmpegsource2 (which I can't do on all sources since it's not frame-accurate with VOBs).

Quote:
If you want to roll your own avisynth.dll without audio cacheing. Edit src\core\cache.cpp look for Cache::GetAudio. After the range checking (before zero, after end of clip) and protection code add a return child->GetAudio(start, count, env); Good luck.
Thanks!

David
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Old 15th August 2012, 19:46   #14  |  Link
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Quote:
Originally Posted by davidhorman View Post
Ah, so it could be that the non-linear access is confusing nicac3source,...
And all decoders.

If I have a Test.ac3 and split the ac3 in two, Test1.ac3 (frames 0 to 284, 285 frames at 32 ms -> 9120 ms) and Test2.ac3 (the rest), and decode the full stream to wav and the two parts you can see in the attached image a glitch (5 ms) similar to the image in your first post.

To decode a frame correctly the decoder need a initilization value from previous frame, if isn't present the first 5 ms are incorrect and can produce audible glitches.

If AviSynth request a sample exactly a 9120-9125 ms, and the previous decoded frame isn't the 284, the decoder isn't correctly initialized to decode the frame 285.

I don't know for what you need manage video+audio with many Trims, but I recommend you decode the full ac3 to wav before and use RaWavSource to add the audio.
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Old 15th August 2012, 21:26   #15  |  Link
mastrboy
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a workaround might be using split_aud: http://mod16.org/hurfdurf/?p=33 and muxing in the output from that later instead of encoding the audio within avisynth script containing trim()

edit: almost forgot, https://github.com/wiiaboo/vfr/ is a python alternative to split_aud
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Old 15th August 2012, 21:51   #16  |  Link
wonkey_monkey
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Quote:
Originally Posted by tebasuna51 View Post
And all decoders.

I don't know for what you need manage video+audio with many Trims, but I recommend you decode the full ac3 to wav before and use RaWavSource to add the audio.
Thanks for the idea - but as my sources amount to over 60 hours of video and my script has almost 1000 trims it may not work in this case

It seems that my earlier problems with EnsureVBCMP3Sync are no longer occurring (previously it wouldn't play back at all unless I had audiograph in my script), and that seems to be solving the problem - just not sure how much it will slow down my encoding yet!

David
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Old 15th August 2012, 23:06   #17  |  Link
IanB
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@tebasuna51,

Would a preroll decoding of the previous frame when ever you seek out of order get rid of the 5ms glitch or is the dependency more complicated?
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Old 16th August 2012, 08:15   #18  |  Link
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Quote:
Originally Posted by davidhorman View Post
Thanks for the idea - but as my sources amount to over 60 hours of video and my script has almost 1000 trims it may not work in this case
60 hours! Well, if you have enough disk space is not a problem.
RaWavSource can support WAV files greater than 4 GB or W64 files.
You can use eac3to or BeHappy to decode AC3 to W64.

Quote:
Originally Posted by IanB View Post
Would a preroll decoding of the previous frame when ever you seek out of order get rid of the 5ms glitch or is the dependency more complicated?
Maybe. I will see.
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Old 16th August 2012, 09:06   #19  |  Link
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Quote:
Originally Posted by IanB View Post
@tebasuna51,

Would a preroll decoding of the previous frame when ever you seek out of order get rid of the 5ms glitch or is the dependency more complicated?
That was going to be my next question - all of my sources run through a function to add a timecode, so if I added to that a custom filter which automatically fetches frame n-1 (or at least, the audio) when frame n is requested, would that likely solve the problem?

Would such a filter be able to keep track over all its instances of which frame from which source was previously decoded? Or can it only do so per instance?

Quote:
Originally Posted by tebasuna51 View Post
60 hours! Well, if you have enough disk space is not a problem.
I don't

David
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Old 16th August 2012, 19:48   #20  |  Link
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Preliminary results suggest that this filter solves the problem!

Code:
#include "stdio.h"
#include "windows.h"
#include "avisynth.h"

class preroll : public GenericVideoFilter {   
public:
  preroll(PClip _child, IScriptEnvironment* env);
  void __stdcall GetAudio(void* buf, __int64 start, __int64 count, IScriptEnvironment* env);
};

preroll::preroll(PClip _child, IScriptEnvironment* env) : GenericVideoFilter(_child) {
}

void __stdcall preroll::GetAudio(void* buf, __int64 start, __int64 count, IScriptEnvironment* env) {
  if (start!=0) child->GetAudio(buf, start-count, count, env); // if this isn't the first frame, get the audio for the previous one
  child->GetAudio(buf, start, count, env); // now get the audio for the current frame
}

AVSValue __cdecl Create_preroll(AVSValue args, void* user_data, IScriptEnvironment* env) {
  return new preroll(args[0].AsClip(),env);
}

extern "C" __declspec(dllexport) const char* __stdcall AvisynthPluginInit2(IScriptEnvironment* env) {
  env->AddFunction("preroll", "c", Create_preroll, 0);
  return "`preroll' preroll plugin";
}
This is very rough - I've never written an audio filter before, and don't quite understand the principles yet. So if anyone more experienced could cast a critical eye over the code or test it out (preroll.zip), it would be greatly appreciated.

To do: skip the pre-fetch if that previous frame was the last "current" frame (may not be possible with multiple instances of the filter running)

David
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