Hi, please respect for cats
I cannot understand when I use the AAC encoder instead of the AC3, the audio bitrate becomes alwais variable . If possible I would like only constant and high bitrate
Is there a particular setting of ffmpeg to give constant bitrate using AAC?
This is my batch
https://www.dropbox.com/s/6g8s9eti1qawn5s/OUT_matrix.bat?dl=0
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We had this topic few months ago - seem by design AAC is Variable Bit Rate Audio CODEC. You can search for info which format of the AAC can be filled with stuffing to deliver CBR.
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Also, I have no respect for cats beyond those of other animals. They certainly have no respect for me!
Scott -
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does ffmpeg have another built-in aac encoder?
Note that due to license collisions you need to manually build such a build (for example by using MBAS), builds from zeranoe&co can't incorporate libfdk_aac and similar codecs. (read more details in the link above)
to preserve audio quality
If you need such high bit rates and you want to preserve audio quality aac is the wrong audio codec, ALAC (Apple Lossless Audio Codec) might be more suited.users currently on my ignore list: deadrats, Stears555 -
The POINT/DEFINITION of vbr is to have quality while not having to waste bitrate when it's unnecessary (or vice-versa).
And AAC is a format which has supported vbr from its inception, so most all well-designed apps/devices should fully support vbr in AAC.
Scott -
AAC by design is VBR. Check AAC specification. Or accept this explanation:
https://hydrogenaud.io/index.php/topic,82865.msg718111.html#msg718111
(AAC) does not make a specification of packet sizes, but instead specifies an Average bitrate that should be achieved (ABR) -
in my batch I have set
Code:-c:a ac3 -b:a 320k
ffmpeg tell that "too many bits, ..., clamping to max"
The resultat output is VBR 427 - 576 kB that theorically - in abstract - is not a bad quality.
My source is alwais an analog sampled source of audio that comes from analog receiver sennheiser ek3042 with mirophones capsules SKM-5200/SKM3072 or SK50 with MKE-40 capsules or mke-2, or sennheiser MKH-60 and MKH 816. Audio recording is sampled to 24 bit 48khz, edited scaling to 16 bit and final encoding with ffmpeg at 16bit, 48 khz. Using AAC ( -c:a aac -b:a 320k ) the real bitrate variable become 250 and it seems slightly audible a drop in quality that instead does not occur using AC3Last edited by marcorocchini; 16th Feb 2019 at 14:50.
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Both are lossy. Both can be set low bitrate & low quality, or with high enough bitrates so as to be transparent...to most users. If you are using these as source files to be sent elsewhere to be further worked on, if I were them, I would be pretty pissed off.
Scott
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