![]() ![]() (…) But would transcoding the opus to AAC (…) maintain that quality difference, or would the transcoding (…) induce artefacts so that the audio quality (…) would actually be worse? I guess you won't.Īnd finally, youtube-dl also can download an AAC file of 128 kbs. You may even use -vbr 4 and check whether you can detect the difference. This should be absolutely transparent to anyone's ears. Thus, the following would work: ffmpeg -i input.opus -c:a libfdk_aac -vbr 5 -cutoff 18000 output.m4a turning off the high pass filter, as by default some high frequencies are lost during conversion.using VBR - there is no need to constrain the rate or waste bits. ![]() ![]() using the libfdk_aac encoder instead of the built-in one (the encoder can make a huge difference in quality!).Here you will find tips on how to achieve the best quality, which involves: More generally, I would recommend to have a look at the FFmpeg Wiki page on AAC encoding. The first answer given seems to suggest this. The resulting file would in any case not be larger than the 5Mb, as I assume no new information can be invented/generated in the process (I will do some experimenting over the weekend to test this). So is my understanding correct that the maximum information in the file is not more than 5Mb, so to encode (and if it is not a requirement to reduce file size) I should just use the highest possible quality VBR, or possibly a high CBR (256 or 320) to capture virtually all information still present in the opus file. The first step in transcoding is to first decode the file (to a PCM stream?), and then encode again in the new format. highest settings for lame or qaac), and 7-8Mb for a 320kbs CBR).īut in this case, the source file is a 5Mb 128kbs opus file. For arguments sake, let’s say maybe 3Mb for a 128kbs MP3, 5-6Mb for a top quality VBR ( i.e. The encoder then does its hocus pocus (discard inaudible info, compression, etc.), which then give you a smaller lossy file, based on the settings you choose. a 50Mb wav), and you choose encoder settings for quality (VBR) or desired bitrate (CBR/ABR). In most cases, you start with lossless file (e.g. Let me explain why I “conceptually” struggle with understanding how to determine optimized settings for transcoding. at 192kbs/212kbs) maintain that quality difference, or would the transcoding (even to a higher bitrate) induce artefacts so that the audio quality of the 192kbs AAC file would actually be worse than the AAC 128kbs file ? EDIT But would transcoding the opus to AAC (e.g. Given that opus is more efficient/better, quality will be higher at the same bitrate. I know that from CD or lossless, VBR is most recommended, but for transcoding from Opus, would CBR be better?Īnd should I keep the 48khz sample rate from the opus file, or downsample to the usual 44.1khz of AAC?Īnd finally, youtube-dl also can download an AAC file of 128 kbs. Would 192kbs be a good choice for the AAC file? (I presume AAC is in any case a better choice than MP3?) )?Īccording to the youtube-dl -F data it is 160kbs, but when downloaded, MediaInfo gives 127Kbps overall bitrate and 48Khz sampling rate. ffmpeg) would you recommend to convert an opus file to AAC to keep same audio quality (I know this is subjective. I have read here that you can mitigate the generation loss to a certain extent by using a higher bitrate for the target codec. using same container, across Windows/iOs/car, I want to convert to AAC or MP3. However, as I want to use the same file, i.e. I know that (1) Opus is the newest and more efficient codec, and (2) converting from one lossy codec to another is not recommended. I am downloading some music from Youtube, and it seems that in most cases (popular videos), the best quality audio is an opus file.
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