![]() ![]() Simply download the plugin 3, copy the contents of the zipfile in your main folder of VirtualDub and (re)launch VirtualDub and you'll be good to go! The only thing you need is a plugin called Virtualdub FFMpeg Input Plugin 2. specification), when the actual problem lies in one specific broken/outdated/incomplete software implementation of that format.VirtualDub on its own sadly doesn't support many popular codecs and formats, but there is a way to open your favorite mp4 or mkv (and many many others). ![]() And it obviously must be able to deal with that fact.Ĭonsequently you must not blame the container format (i.e. Software that does supports HE-AAC must be aware that the final decoded + post-processed output sample-rate may be different from the "base" rate. That is exactly the sample rate that a Non-HE-capable AAC decoder would return. Hence it only specifies the the file format, but it does not specify how the software that reads the file has to behave.įor HE-AAC it makes sense to specify the "base" (not re-upconverted) sample-rate as the stream's sample-rate in the AVI header. VFW is outdated/discontinued, as we all know.Īt the same time AVI is only a container format. But VFW is only one (out of many) specific software implementation to read AVI files. If the AVI audio header declares that the sample rate is 22050 Hz (as this one does), then that is indeed what you will hear. I have no control over the sample rate in an AVI file. ![]() (Have a look at "MP3-VBR-in-AVI is a dirty hack" and "AAC in AVI does not work") Whether the AAC decoder, which will be used to decompress the AAC data later, does support HE-AAC (SBR) or not, that is a completely different and AVI-unrelated question. Only some buggy AVI splitters don't handle it properly, because they make the (false) assumption that audio streams will always use fixed-size samples. Audio streams with varying sample size work just fine with AVI - if done in the right way. Still that is a property of the AAC format and is not related to AVI at all. When using an HE-AAC-capable decoder, the output will be upsampled to the original sample rate and the "missing" frequencies will be synthesized. And indeed it will divide the the sampling rate by 2 (and thus "remove" the higher frequencies according to Nyquist Theorem).Īs a result, you will get only 1/2 sampling rate (and thus worse quality) compared to the original, when using an AAC decoder that does not support HE-AAC (SBR). SBR is pre-processing before the actual AAC encoding step. ![]() Actually SBR can be applied to MP3 just as well (known as "MP3pro"). The format of the resulting AAC stream is no different. ![]()
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