I haven't tried decoding but the Nero encoder won't encode 88.2kHz audio as HE-AAC, only LC-AAC. PS I wouldn't hold my breath trying to get the Nero decoder to decode it. And check the MKA with MediaInfo to see if it's still reporting the same sample rates. Try converting the MKA to a wave file to see if that makes a difference. You can open it with MKVToolNixGUI, select the AAC stream, and under the audio and subtitles tab, change the AAC SBR dropdown box to "yes", then save that as an MKA. Like this:Īnd maybe try putting the raw aac into a container before converting it in case it's not being identified correctly. Add -report to the ffmpeg command line and see if the log file it produces sheds any light on the subject. I converted with ffmpeg and foobar2000 and I also extracted the raw AAC and converted that too, thinking maybe that's the problem, but it was still fine.Ī couple of suggestions. I tried to duplicate the problem but no matter what I did I ended up with a 88.2kHz wave file.
FFMPEG 4 BIT PCM TO WAV FULL
To make sure I'm not full of crap I made a similar HE-AAC stream and converted it. any AAC decoder should be able to decode it properly. It's supposed to be backwards compatible so a player supporting only LC-AAC can still decode it, ignoring the SBR part, so it'll sound a bit dull as the high frequencies aren't included.Īnyway. 32 bit depth x 48,000 sampling frequency 1,536,000 bits per second. That is because the only bit depth that codec supports is 32 bits per sample. HE-AAC uses spectral band replication which "fakes" the high frequency stuff on playback by creating some sort of guide track to do it, at half the original rate. Presuming ffmpegs default AC-3 encoder, if you simply set the sample rate to 48kbps (which you did in your example above), youd be encoding at 1,536kbps for all channels. The audio is sampled at the original rate (88.2k in this case) but the high frequencies are removed. > that's one of the main reasons why he-aac can produce decent output on such low bitrates. ffmpeg -t option can now be used for inputs, to limit the duration of data read from an input file - incomplete Voxware MetaSound decoder - read EXIF metadata from JPEG - DVB teletext decoder - phase filter ported from libmpcodecs - w3fdif filter - Opus support in Matroska - FFV1 version 1. wav: ffmpeg -i video.flv -c:a pcms16le audio-only.wav: Encode to DNxHR-HQ and PCM audio: ffmpeg -i video.mov -c:v dnxhd -profile:v dnxhrhq -pixfmt yuv422p -c:a pcms16le video-dnxhr-gq-raw-pcm. No, when you encode 88.2kHz content with a he-aac encoder it only saves information + some data for reconstruction and the job ob the decoder is to reconstruct the original 88.2kHz sample rate. FFMpeg command Extract audio from video to raw 16-bit PCM.
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